Download IP Phone User's Guide

SETU VP236
System Manual
Magyarországon a Matrix Telecom Ltd. képviselete,
Matrix termékek importőre, kizárólagos forgalmazója:
1095 Budapest, Mester u. 34.
Telefon: *218-5542, 215-9771, 215-7550, 216-7017, 216-7018
Fax: 218-5542 Mobil: 30 940-1970, 20 949-2688
E-mail: [email protected] Web: www.delton.hu
Documentation Information
This is a general documentation and it covers many models with
different specifications. A particular product may not support all the
features and facilities described in the documentation.
Matrix Telecom reserves the right to revise information in this
publication for any reason without prior notice. Information in this
documentation may change from time to time. Matrix Telecom makes
no warranties with respect to this documentation and disclaims any
implied warranties. While every precaution has been taken in
preparation of this user’s guide, Matrix Telecom assumes no
responsibility for errors or omissions. Neither is any liability assumed
for damages resulting from the use of the information contained
herein.
Matrix Telecom reserves the right without prior notice to make
changes in design or components of the equipment as engineering
and manufacturing may warrant.
Neither Matrix Telecom not its affiliates shall be liable to the purchaser
of this product or third parties for damages, losses, costs or expenses
incurred by purchaser or third parties as a result of: accident, misuse
or abuse of this product or unauthorized modifications, repairs or
alterations to this product or failure to strictly comply with Matrix
Telecom’s operating and maintenance instructions.
All rights reserved. No part of this user’s guide may be copied or
reproduced in any form or by any means without the prior written
consent of Matrix Telecom.
Contents
Section 1: Introduction ............................................................... 5
Welcome ....................................................................................... 7
Packing List .................................................................................. 9
Warranty Statement .................................................................... 10
Care for Your VP236 ................................................................... 13
Introducing the System ............................................................... 14
Configuring SETU VP236 ........................................................... 16
Know Your SETU VP236 ............................................................ 17
Getting Started ............................................................................ 27
Section 2: Features and Facilities ........................................... 35
3-Party Conference .....................................................................
Anonymous Call Rejection ..........................................................
Auto Answer ................................................................................
Auto Call Back ............................................................................
Auto Configuration ......................................................................
Black Listed Caller’s List .............................................................
Call Appearances ........................................................................
Call Forward ...............................................................................
Call Hold .....................................................................................
Call Logs .....................................................................................
Call Mute .....................................................................................
Call Progress Tone Generation (CPTG) ......................................
Call Toggle ..................................................................................
Call Transfer ...............................................................................
Call Waiting .................................................................................
Change Password ......................................................................
CLIR ............................................................................................
Configuring the Phone-Menu ......................................................
Daylight Saving Time ..................................................................
Default SETU VP236 ..................................................................
Dial Plan .....................................................................................
Do Not Disturb (DND) .................................................................
Hotline .........................................................................................
Key Assignment ..........................................................................
Keypad Lock ...............................................................................
37
39
40
42
44
47
49
52
56
60
63
65
68
69
72
73
74
75
76
79
81
83
85
87
88
Last Number Redial .................................................................... 91
LCD Settings ............................................................................... 92
Making a Call .............................................................................. 94
Making Second Call .................................................................... 99
Menu ......................................................................................... 100
Network Port ............................................................................. 102
Notifications .............................................................................. 108
OG Call Route Selection ............................................................ 110
PC Port ...................................................................................... 112
Peer-to-Peer Calling .................................................................. 113
Phone Book ............................................................................... 117
Receiving a Call ......................................................................... 119
Rejecting a Call ......................................................................... 122
Rings ......................................................................................... 124
Search Contact ......................................................................... 126
Selective SIP Line ..................................................................... 127
SIP Parameters ......................................................................... 128
Software Upgrade ..................................................................... 134
Speed Dial ................................................................................ 136
Status ........................................................................................ 138
Switching between the Handset, Headset and Speaker ........... 142
System Parameters .................................................................. 144
Time Settings ............................................................................ 148
Timers ....................................................................................... 150
User Settings ............................................................................ 152
VLAN Tagging ........................................................................... 154
Voice Mail ................................................................................. 157
Volume Settings ........................................................................ 160
VoIP Basics ............................................................................... 162
Web Jeeves .............................................................................. 168
Section 3: Appendices ............................................................ 177
Appendix A: Glossary ............................................................... 179
Appendix B: Product Specifications .......................................... 182
Index ........................................................................................ 185
Section 1: Introduction
Matrix
Welcome
Thanks for purchasing a Matrix product.
We want you to get the maximum performance from our product. If
you run into technical difficulties, we are here to help. But please
consult this system manual first.
If you still can’t find the answer, gather all the information or questions
that apply to your problem and with the product close to you, call your
dealer. Matrix dealers are trained and ready to give you the support
you need to get the most from your Matrix product. In fact, most
problems reported are minor and can be easily solved over the phone.
In addition, technical consultation is available from Matrix engineers
every business day. We are always ready to give advice on
application requirement or specific information on installation and
operation of our products.
The system manual is divided in following sections:
Section 1: Introduction
Section 2: Features and Facilities
Section 3: Appendices
We suggest the first time users to read this system manual in the
following sequence:
• Section 1
• Section 2 (In the sequence given below for few topics)
• Web Jeeves 168
• Menu 100
• Status 138
• Configuring the Phone Menu 75
• Key Assignment 87
• Auto Configuration 44
• Change Password 73
• Call Progress Tone Generation 65
• LCD Settings 92
• Network Port 102
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Matrix
•
•
•
•
•
•
8
Time Settings 148
OG Call Route Selection
Making a Call 94
Receiving a Call 119
SIP Parameters 128
Call Logs 60
110
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Packing List
• The ideal sales package contains:
Sr.
Item
01 SETU VP236S/P
02 System Manual
03 Handset
04 Adaptor 5V, 2.0 Amp.
05 RJ45 Cable
06 4P4C Spring Cable
07 Wall Mount Bracket
08 Warranty Card Set
09 Support Card
10 Telephone Message Pad
11 SETU VP236 CD
Qty.
1
1
1
1
1
1
1
1
1
1
1
• Please make sure that these components are present.
• In case of short supply or damage detection, contact the source
from where you have purchased the system.
=X=X=
SETU VP236 V1 System Manual
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Matrix
Warranty Statement
Matrix Telecom Pvt. Ltd. (Matrix) warrants to its consumer purchaser
any of its products to be free of defects in material, workmanship and
performance for a period of 15 months from date of manufacturing or
12 months from the date of installation which ever is earlier.
During this warranty period, Matrix will at its option, repair or replace
the product at no additional charge if the product is found to have
manufacturing defect. Any replacement product or part/s may be
furnished on an exchange basis, which shall be new or like-new,
provided that it has functionality at least equal to that of the product,
being replaced. All replacement parts and products will be the property
of Matrix. Parts repaired or replaced will be under warranty throughout
the remainder of the original warranty period only.
This limited warranty does not apply to:
1. Products that have been subjected to abuse, accident, natural
disaster, misuse, modification, tampering, faulty installation, lack
of reasonable care, repair or service in any way that is not
contemplated in the documentation for the product, or if the model
or serial number has been altered, tampered with, defaced or
removed.
2. Products which have been damaged by lightning storms, water or
power surges or which have been neglected, altered, used for a
purpose other than the one for which they were manufactured,
repaired by customer or any party without Matrix’s written
authorization or used in any manner inconsistent with Matrix’s
instructions.
3. Products received improperly packed or physically damaged.
4. Products damaged due to operation of product outside the
products’ specifications or use without designated protections.
Warranty valid only if:
• Primary protection on all the ports are provided
• Mains supply is within limit and protected
• Environment conditions are maintained as per the product
specifications
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Warranty Card:
• When the product is installed, please return the warranty card
with:
• Date, signature and stamp of the customer.
• Date, signature and stamp of the channel partner.
• Matrix assumes that the customer agrees with the warranty
terms even when the warranty card is not signed and returned as
suggested.
The Purchaser shall have to bear shipping charges for sending
product to Matrix for testing/rectification. The product shall be shipped
to the Purchaser at no-charge if the material is found to be under
warranty. The Purchaser shall have to either insure the product or
assume liability for loss or damage during transit.
Matrix reserves the right to waive off or make any changes in its
warranty policy without giving any notice.
If Matrix is unable to repair or replace, as applicable, a defective
product which is covered by Matrix warranty, Matrix shall, within a
reasonable time after being notified of the defect, refund the purchase
price of the product provided the consumer/purchaser returns the
product to Matrix.
In no event will Matrix be liable for any damages including lost profits,
lost business, lost savings, downtime or delay, labor, repair or material
cost, injury to person, property or other incidental or consequential
damages arising out of use of or inability to use such product, even if
Matrix has been advised of the possibility of such damages or losses
or for any claim by any other party.
Except for the obligations specifically set forth in this Warranty Policy
Statement, in no event shall Matrix be liable for any direct, indirect,
special, incidental or consequential damages whether based on
contract or any other legal theory and where advised of the possibility
of such damages.
Neither Matrix nor any of its distributors, dealers or sub-dealers makes
any other warranty of any kind, whether expressed or implied, with
SETU VP236 V1 System Manual
11
Matrix
respect to Matrix products. Matrix and its distributors, dealers or subdealers specifically disclaim the implied warranties of merchantability
and fitness for a particular purpose.
This warranty is not transferable and applies only to the original
consumer purchaser of the Product. Warranty shall be void if the
warranty card is not completed and registered with Matrix within 30
days of installation.
All legal course of action subjected to Vadodara (Gujarat, India)
Jurisdiction only.
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SETU VP236 V1 System Manual
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Care for Your VP236
Power
• To disconnect power from the device the electric plug and socket
must be easily accessible at all times.
• The standard power adaptor is supplied with a Phone. The phone
should only be operated with this universal power adaptor.
Damages to the phone caused by using other power adaptors are
not covered by the manufacturer’s warranty.
• This phone can be powered if Power on Ethernet (PoE) is
available, without using Adaptor.
Shock Hazard
• Avoid operating this device during periods of lightning.
• Insert carefully the adaptor in the 230VAC socket.
Protection from Water, Heat, Sun and Cold
• Do not expose the phone to water, as this will create a shock
hazard.
• The phone is an electronic device and should not be exposed to
excessive heat, sun or cold.
Cleaning Your VP236 Phone
• Please use a lightly water moistened paper or cloth towel.
• Do not spray or pour cleaning solution directly onto the phone.
=X=X=
SETU VP236 V1 System Manual
13
Matrix
Introducing the System
• Thank you for purchasing Matrix SETU VP236. You made an
excellent choice and we hope you will enjoy all its capabilities.
• The SETU VP236 is a SIP based VoIP phone. Sending voice
signals over Internet is called Voice Over IP (VoIP). And Session
Initiated Protocol (SIP) is an internationally recognized standard for
implementing VoIP.
• This SIP SETU VP236 is the innovative enterprise IP telephone
that offers a rich set of functionality and superb sound quality. They
are fully compatible with SIP industry standard and can
inter operate with many other SIP compliant devices and software
available in the market.
• These specialized phones look just like normal phones with a
handset, cradle and buttons. But instead of having the standard
RJ-11 phone connector, SETU VP236 have an RJ-45 Ethernet
connector. It can be connected directly to router and have all the
required hardware and software right onboard to handle the IP call.
• Dialing a number from the phone and Receiving the call is just like
normal phone. Operating different features is user friendly.
• LCD display gives information about SIP account and other details
regarding active call and features set. It can also display CLI of
Incoming call.
• All important features are programmable through Web pages as
well as features in Menu on the Phone. Some call features can be
assigned to the keys on the phone so that features can be directly
used. For this the phone has a key pad with 36-keys. Some keys
are with LED’s also for the indication of features set currently.
• Matrix SETU VP236 supports maximum of 2-SIP Accounts, and
4-Call Appearances.
• Some terminology used in this Manual are described as below:
• User: The person who is using the SETU VP236.
• Remote User: The person with whom the user wants to interact/
interacts.
• Transfer Target: The remote user to whom the call is to be
transferred.
• Transferor: The user who transfers the call.
• Transferee: The user who is to be transferred.
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SETU VP236 V1 System Manual
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• Line: The SIP1 or SIP2 Trunk.
The words SETU VP236, IP Phone, Phone and system are used
interchangeably in this manual.
Before proceeding further, please ensure that you have an
Internet access and a SIP Account already set up.
Relevant Topics:
1. VoIP Basics 162
2. Glossary 179
=X=X=
SETU VP236 V1 System Manual
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Matrix
Configuring SETU VP236
• Generally user will come across three varieties, while programming
different features using the Matrix IP-Phone.
• Accordingly these can be grouped as mentioned below:
• One-time programmable parameters (e.g. Ring Type, CPTG
Type, etc.).
• Less Frequently Used parameters (e.g. DND List, Black Callers’
List, etc.).
• Frequently Used parameters (e.g. Set/cancel Call Mute, Set/
Cancel DND, etc.).
• The parameters which need to be programmed once only, can be
programmed through Web Interface only.
• Less frequently used parameters like Auto Answer Timer, etc. can
be programmed both through Web as well as Phone. Programming
through Phone can be done using an option called ‘Configuring the
Phone’ provided in the Menu of the Phone.
• Frequently used parameters like Set/Cancel Hotline are settable
from Web as well as from Phone. Programming through Phone in
this case is directly through Menu of the Phone.
Programming
Using WEB
Phone Menu
Using Phone ‘Menu’
Using feature ‘Configuring the Phone’
Refer chapter ‘Know Your SETU VP236’ for list of following
parameters:
• Features programmable from ‘Configuring Phone Menu’.
• Features programmable from ‘Menu’.
• Features which can be assigned a ‘Key’.
=X=X=
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SETU VP236 V1 System Manual
Matrix
Know Your SETU VP236
Matrix SETU VP236 has the facility to use keys and LCD as explained
in following sections:
• LCD Display
• Keypad and Ringer LED
• Key assignment
• Function specific key
• Using Menu key
• Options of feature ‘Configuring the Phone’
• Using Dial pad key Interface
LCD Display:
Matrix supports two varieties of SETU VP236:
• SETU VP236S (2x24 Line LCD)
• SETU VP236P (6x24 Line LCD)
• These Phones support similar features and can be used any one
directly without any change. The only difference is the display
capacity. Type-S comes with 2-line, 24-characters and Type-P
comes with 6-line, 24-characters LCD display.
• While dialing a number, all 24 characters of LCD display are used
to display the number string. First 18 characters are used to display
the number string, 19th character is blank and last five characters
are used to display the call progress status.
• User can dial number string with maximum of 40 characters. If
dialed number string has more then 40 char, the phone will dial only
first 40 char of number string.
• User should program Max. 24 characters from the phone
configuration even if the maximum field length allowed for the
number string/ address is more then 24.
• User should configure the Number string/ address having more
then 24 characters using Web Jeeves.
• Display strings are same for 2-line LCD display and 6-line LCD
display. The content of the strings (text messages) are so selected
that single line is sufficient to convey the message.
SETU VP236 V1 System Manual
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Matrix
For Example:
• If the user enters wrong password, the display string selected is
‘Invalid P/w, Try again’.
• Display pattern are also same for 2-line LCD display and 6-line
LCD display.
• For the normal display which consists of 6 lines the content of
‘display’ is as described below:
• First Line:
It will display Menu and Day, Date and Time which is not affected
by Incoming or Outgoing call.
• Second Line to Fifth Line:
It is used to display status of CA1 to CA4 (Call Appearance).
When a CA is free, a text string ‘CALLx Free’ is displayed.
• The CA row on the LCD can be treated as CA key. Doing so,
following is possible:
• In idle state, if the user presses ‘Enter’ key on any of the CA row,
it is treated as OFF-Hook. If speaker key is pressed, user gets
dial tone and ‘Dial….’ is displayed, in place of ‘CALLx free’
• While talking to remote user if the user presses ‘Enter’ key on the
CA row on which the current call is going through, the call is held.
The display will now show the dialed number/name and a string
‘Hold’ in the call status characters.
• Likewise for all other functions which are possible from the CA
keys are possible from CA rows.
• Sixth Line:
It displays notification/feature settings. These strings are displayed
only when the notification is received or a feature is set. Otherwise
that character will display space (blank).
• The 2-line LCD user can just scroll, to know the status of other CA
as only the first CA is displayed on the second line of the LCD and
can select the option by pressing enter key. Whereas the 6-line
LCD user will be able to view all the 6 lines of the normal display.
• Idle Condition:
• Normal parameter display on Phone will be as shown below:
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SETU VP236 V1 System Manual
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• When the phone is in idle state the LCD display will now be
displaying all the Call Appearances Free as shown above.
• When Menu is accessed, all the features are displayed in
alphabetical order. Refer topic ‘Menu’ for more details.
• When the user makes a selection, the feature title is displayed in
the first row whereas its options are displayed in subsequent rows.
For Example:
• When the user selects ‘Call Forward-Busy’, CALL FORWARDBUSY is displayed in the first row whereas Set, Cancel options are
displayed in subsequent rows.
Keypad and Ringer LED:
• Matrix SETU VP236 supports a keypad with dial pad keys,
programmable keys, Speaker/Headset key and feature keys as
explained below:
• 12 Dial pad keys (Non-programmable)-Fixed keys
• 5 Navigational keys (Non-programmable)
• 1 Speaker key (Non-Programmable) with ‘Red’ LED for speaker
ON indication.
• 18 feature keys (Programmable)-Keys 06 to 09 have dual color
LEDs (Green and Red), keys 01, 02 and 10 to 13 have single
color LEDs (Red) whereas remaining keys do not have a LED.
• Ringer LED (Single Color) is provided on right-top side of the
phone to indicate the IC call.
• Keypad can be locked.
SETU VP236 V1 System Manual
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Matrix
• Default, Key Assignment is as shown in diagram:
Key
01
Label
VM Account1
Label
Mute
Key
10
02
CFB
11
03
CFNR
12
04
DND
13
Reject
14
05
CA 4
06
Redial
15
07
CA 3
Conference
16
08
CA 2
Transfer
17
09
CA 1
Ringer LED
18
1
2
3
4
5
6
7
8
9
*
0
#
Spk
Key assignment:
• Matrix IP Phone provides the feature, “Key Assignment”.
• With this feature, user can access the feature just by pressing the
key assigned.
For Example:
If a user uses a feature ‘Hot Line’ feature, repeatedly and he wishes to
use the programmable key for this purpose, then this feature is useful.
User can label that key and use every time.
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SETU VP236 V1 System Manual
Matrix
• Table showing features which can be assigned to programmable
key, using web page:
Feature to which key
Any LED Activity
canbe assigned
on the key?
None
N
Alphanumeric Dialing
Y
Auto Answer
Y
Auto Call Back
Y
Call Appearance 1
Y (Dual Colour)
Call Appearance 2
Y (Dual Colour)
Call Appearance 3
Y (Dual Colour)
Call Appearance 4
Y (Dual Colour)
Call Forward-Busy
Y
Call Forward-No Reply
Y
Call Hold
N
Call Logs
N
Call Mute
Y
Call Transfer
N
CLIR
Y
Conference
Y
DND-Set/Cancel
Y
Headset
N
Hotline-Set/Cancel
Y
Keypad Lock
Y
Last Number Redial
N
Menu
N
Phone Book (Speed Dial)
N
Reject
N
Search Contact
N
SIP1
Y
SIP2
Y
VM Account 1
Y
VM Account 2
Y
SETU VP236 V1 System Manual
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Matrix
• Key assignment can be done using Web only.
• Please note that features needing dual color LEDs (like Call
Appearance) should be assigned keys 06 to 09. Likewise features
needing single color LEDs should be assigned keys 01, 02 or 1013. Whereas features not needing a LED should be assigned to a
key with no LED.
BE CAREFUL TO SELECT A KEY WITH LED OF REQUIRED NUMBER
OF COLOURS FOR A FEATURE WHICH IS TO BE ASSIGNED.
• ‘None’ option is to indicate that key is not assigned a feature.
The Function Specific Keys:
spk :
This key is fixed for ‘Speaker’ mode.
‘<’ :
This key is used to decrease the Rx Gain of Call Progress
Tones in all port states (Dial, Speech, etc.) or to go to Main
Menu from Sub-menu.
‘>’ :
This key is used to increase the Rx Gain of Call Progress
Tones in all port states (Dial, Speech, etc.).
‘√’ :
This key is used as selection or Enter key.
This key is used to navigate parameters of ‘Menu’ to upward.
‘∧’ :
This key is used to navigate parameters of ‘Menu’ to
‘∨ ’ :
downward.
Using Menu key:
• User can access certain features using ‘Menu’ feature if that
feature is not assigned a key. For example ‘CLIR’ can be enabled
from the Phone just by navigating the ‘Menu’ ans selecting ‘CLIR’.
Menu can be accessed by following ways:
• Normal Display on the LCD. Select the line displaying ‘Menu’
and press ‘Enter’ key.
• Using the specific key assigned for ‘Menu’. The key assignment
can be done only from the Web-Page.
• For more information refer Chapter “Menu”.
• Following features can be accessed by ‘Menu’ key.
• Access SIP1 trunk
• Access SIP2 trunk
• Alphanumeric dialing
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•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Auto Answer
Auto Call Back
Call forward-busy
Call forward-no reply
Call hold
Call logs
Call mute
Call transfer
Call VM1 Server
Call VM2 Server
CLIR
Conference
Configuring the Phone
Do Not Disturb
Headset-OnHook/OffHook
Hotline
Keypad Lock
Last Number Redial
Network Status
Rejecting a Call
Search Contact
Options of the feature: ‘Configuring the Phone’ of the Menu key:
• User will not have to enter Web Jeeves every time for configuring
the Phone. As a short cut it is possible to enter ‘Config Phone’ by
navigating from Menu feature and add or modify SIP and other
required parameters.
• User can select this option to access ‘not frequently used’ features.
• Thus, using the ‘Config. Phone’ menu, it is possible to program few
parameters of the Phone from Phone itself.
• The feature ‘Configuring the Phone’ is one of the options of the
Menu.
• Phone will ask for the ‘Programming Password’ (PW), when
entering this option.
Remember that this Config. Phone Password is NOT same as
Keypad-Unlock password or Web-Jeeves password.
• Following options are displayed when ‘Configuring Phone Menu’ is
scrolled with up/down key:
SETU VP236 V1 System Manual
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Matrix
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Anonymous Call Rejection
Auto Answer Timer
Black Listed Callers’ List
Call Forward Timer
Change Configuring Password
Change User Password
DND List
Hotline
Keypad Lock Setting
LCD Settings
Network Parameters
Number of Call Appearances
Phone Book
SIP1 Parameters
SIP2 Parameters
Volume Setting
Using Dial Pad key Interface:
• SETU VP236 supports 12 dial pad keys, 18 feature keys, 1
speaker key and 5 navigation keys.
• 12 Dial pad keys and 5 navigation keys are used to dial a number/
SIP ID, enter a number/SIP ID or edit a number/SIP ID.
• By repeatedly pressing various keys, different characters can be
selected as shown in table below: (when you have selected
'Alphanumeric Dialing').
Table of characters:
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Characters selected
Key
First time
No.
pressed
1
2
3
4
5
6
7
8
9
0
*
#
1
2
3
4
5
6
7
8
9
0
Two
times
pressed
.
a
d
g
j
m
p
t
w
+
#
Sp.
…
…
…
…
…
…
@
b
e
h
k
n
q
u
x
-
,
c
f
i
l
o
r
v
y
*
A
D
G
J
M
s
T
z
/
?
B
E
H
K
N
P
U
W
=
!
C
F
I
L
O
Q
V
X
%
’
Nine
times
pressed
:
R
S
Y
Z
·
• User can easily enter non numeric characters as the method of
entering is same as that used by Mobile Phones while writing an
SMS except that the first character in our case is a numeric when
key is pressed ‘first time’.
• Following examples show how to enter/edit a number/SIP ID.
Examples:
• Select first ‘Alphanumeric Dialing’.
• If user presses “1” digit key once, “1” is displayed and after waiting
for 1 second (aprox):
• If user again presses “1” within 1 second, phone will display “.”
and again wait for 1 second.
• If user does not press any key within 1 second or enter other digit
key, an entry of “.” is done and the cursor is moved to the next
position.
• If user presses “2” digit key once, “2” is displayed and wait for 1
second.
• If user does not presses any key within 1 second, an entry of “a”
is done and the cursor is moved to the next position.
• Same method is applicable for digit keys “3”, “4”, “5”, “6”, “7”, “8”,
“9”, “0” and “*” as shown in above Table.
SETU VP236 V1 System Manual
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Matrix
How to delete a character? (Not applicable when dialing)
• If user presses “#” once, “#” is displayed and after waiting for 1
second:
• If user does not press any key, an entry of “#” is displayed and
the cursor is moved to the next position.
• If user again presses “#” key within 1 second, the character is
deleted on that position and the cursor is moved to the next
position.
• For Example, if amit is written, user wants to delete “mit”.
•
Move the cursor to “m”, “amit”
•
Press “#” key twice within 1 second, display will be “a it”.
•
If again “#” is pressed twice within 1 second, display will be “a
t”.
•
If again “#” is pressed twice within in 1 second, display will be
“a “.
How to Override characters?
• The IP Phone supports rewriting the numbers without first deleting
the entire number.
• For Example, if user wants to replace “12345” by “76890”.
•
Move the cursor to “1” as “12345”, now if user enters
“76890”, then, “12345” is replaced by “76890”.
Important Points:
• ‘Menu’ feature can be assigned a key.
• While in Menu if the user does not perform any activity, phone goes
to the normal display mode after 15 seconds (fixed).
• While in option ‘Configuring the Phone’ if the user does not perform
any activity, phone goes to the normal display mode after 60
seconds (fixed).
• When the SETU VP236 prompts for the password and if the user
does not enter the password, phone goes to normal display mode
after 15 seconds.
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Getting Started
Overview
The Matrix SETU VP236 allows you to use Packet Switching
technique of IP network for normal communication at a very low cost
of the call-rate. This is a different technology than circuit switching
deployed by POTS. User needs only a subscribed line of Internet
through a DSL Modem or Router and a PC with Internet Explorer
(V6.0) installed in it.
This chapter gets the SETU VP236 up and running quickly. The
details which we have skipped to make this brief can be found
elsewhere in the user’s guide. It is divided into four sections:
• Getting to know the SETU VP236
• Instruction for connecting the SETU VP236
• Basic steps for configuration
• Making phone calls
Getting to know the Phone
• The Matrix SETU VP236 interfacing ports are located on side panel
and LED’s are on few keys. Both are explained below:
• Connection of ports
• LED’s on Phone keys
Connection of Ports:
Following view shows how the ports are connected to various
interfaces:
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Matrix
Power
Adaptor
(1)
Bottom
Side
(2)
(4)
(3)
(5)
Ethernet
Cable
No. Port Na m e Conne ctor De scription
1
Power
2
Network
Port
3
PC Port
4
Headset
5
Handset
6
MIC
28
DC Jack
Power Adaptor, 5V 2.5Amp. DC. If ethernet
switch supports PoE (Power on Ethernet),
do not connect Adaptor.
RJ45
Connection to cable modem/ADSL modem.
Ethernet activity LED (Green) is glowing. It
is a 10/100 Switch LAN port PoE (802.3agf)
connection is supported, on this port.
It is a 10/100 Switch port for connecting PC
LAN cable connection for co-located PC.
Also connect a cable from SETU VP236 to
RJ45
the PC. You can also use this port as
network port if PoE is not be used.
It is a headset port. To connect head phone
Audio Jack
with suitable impedance.
To connect the handset with speaker and
RJ11
MIC inbuilt.
This is inbuilt port. User can speak to MIC
-for handsfree application.
SETU VP236 V1 System Manual
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LED’s on Phone keys:
• Matrix SETU VP236 supports few keys with single or dual color
LED’s for indication of a feature set e.g. Auto Call Back, DND,
Hotline, etc.
• Keys 06 to 09 : The Dual color LEDs (Green and Red),
• Keys 01, 02 and 10 to 13 : The single color LEDs (Red)
• Ringer LED (Red) : Provided on right-top side of the phone to
indicate the IC call.
The reference number for the keys are shown below:
Keys with LED:
Refer chapter ‘Know your SETU VP236’.
Power ON sequence for the SETU VP236:
• LED sequence during power ON is as shown below:
Cadence (sec.)
Single/Dual
Color LED
ON
OFF
RED
15
0
(Continuous)
Indications on Call Appearance-LED for call under progress:
• The status of the call is displayed on the LEDs embedded in the
respective key assigned to the CA feature.
Call Event
Color
On
Idle
Off
-Dialing
Red (Cont.)
-Proceeding
Red
500ms
Ringing (Far end)
Red –› Green
500ms
Busy
Orange
500ms
Error
Orange (Cont.)
-Connected
Green (Cont.)
-Hold
Green
500ms
Hold Alert (for 5 secs.) Green –› Red
500ms
Remote Hold
Orange –› Red
500ms
Incoming
Orange –› Green 500ms
Far end rejection
Orange
500ms
SETU VP236 V1 System Manual
Cadence
Off
On
----1sec
--500ms
1sec
-----1sec
--500ms
1sec
--500ms
500ms
--
Off
------------
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Matrix
Instruction for connecting the phone:
This section describes the instructions on, how to connect the Matrix
SETU VP236 to internet network.
Unpack the box. Get satisfied with the contents and the condition of
the parts. Refer to ‘Packing List’. If parts are OK, proceed with
connections as shown below:
• Mount your SETU VP236 in a safe and convenient location where
cables for your network and phone system are accessible. For this,
mechanical drawing can be used as a reference, which is provided
at end of the chapter.
• Insert one end of the straight cable into the LAN port of SETU
VP236 and connect the other end of the cable to an uplink port (a
router or a DSL modem), which is used to access internet.
• Insert one end of the cross cable into the PC port of VP236 and
connect other end of the cable to the PC.
• Connect the Audio Pin of the Headphone to the “Headset” port if
you want to use the SETU VP236 in Headset-mode.
• Check the voltage from the power point from where the supply is to
be accessed. It should be between 90-265VAC, 47-63Hz. Insert
the DC output terminal of power adapter into the ‘Power’ jack of the
SETU VP236 and connect the AC Mains pins of the power adapter
to a wall outlet for AC Mains. You can also get power using PoE if it
is supported.
• When you power ON the SETU VP236, booting sequence will start
and observe normal display on the LCD.
Basic steps for configuration
Network Port:
• Default Network IP Address for SETU VP236 is 192.168.001.161
ABSOLUTELY NEEDED !
Use WAN IP Information from your
LAN Admin. or ISP
• IP Address (must)
• Subnet Mask
• Gateway
• DNS IP Address
• DNS Domain Name
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• Note that subnet of WAN port, Internet (Router) and PC are same.
• If not, you will be required to correct the IP address of the PC or of
the SETU VP236.
To correct IP address of the PC:
Read the IP address of PC, by clicking on Network Neighborhood Æ
Properties Æ TCP/IP Connection Æ Properties for Windows98.
• Choose the unique IP address of the PC such that its subnet is
same as subnet of IP address of SETU VP236, provided by ISP.
PLEASE CHECK !
The subnet of your PC IP Address
must be same as of VP236.
• For example, if IP address of VP236 is 192.168.5.10, then PC
IP Address port should be programmed as 192.168.5.X. where ‘X’
can be any number from 1 to 254, except 10. By doing this, subnet
remains same.
• Now if you require to correct the default Static IP Address of the
Phone, proceed as mentioned below:
• Your phone may be with ‘key pad locked’. Then while pressing
the key, the Phone will ask “User Password’. Enter the default
password ‘1111’. If password is correct, the keypad is ‘unlocked’
and you can further operate the Phone.
Remember that this ‘keypad unlock password
is Not same as the one for entering to Web
jeeves from the ‘Welcome’ page.
• Press the ‘Menu’ key (if assigned) of the Phone. If key is not
assigned, select ‘Menu’ from the LCD display and press ‘Enter’
key. Now you are in Menu mode. Navigate the parameters with
up/Down key. Get the Parameter “Configuring the Phone”. After
selecting, when you press the ‘Enter’ key, the Phone will ask the
Password.
Remember that this Config Phone Password is same as
password for entering Web-Jeeves.
• You can navigate the parameters by up/down key after entering
correct Password (Default is 1234). Select ‘Config the Phone’ Menu
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and navigate to ‘SIP Para’. Press ‘Enter’ to get different
programmable parameters for SIP1 account. Select “IP-Address” if
you want to modify it. Press ‘enter’.
• Similarly you can enter to ‘Network Parameters’ and select the
WAN IP Address. For example, if you want to enter IP Address
192.168.1.120. enter digits as: 192111681100111120 (‘11’ is
interpreted as ‘.’ by the Phone).
This is known as Static IP-Address which is enabled by default.
• Now at Powering ON, if Connection type programmed in the Phone
is ‘DHCP’ (DHCP enabled) then System will request following
information from DHCP Server:
• IP Address
• Subnet Mask
• Gateway Address
• DNS Address (If DNS Server is set for DHCP)
• If DHCP Server provides IP Address successfully, the Phone will
use this address instead of Static IP Address.
• Once the SETU VP236 has its own IP address, it shall further try
for SIP registration or auto configuration if programmed. These
parameters can not be edited by the user.
• Similarly if PPPoE is enabled, while powering ON the Phone, then
System will request following Authentication information from
PPPoE Server:
• User ID
• User Password
• Once the SETU VP236 has its own IP address, it shall further try
for SIP registration or auto configuration if programmed. These
parameters can not be edited by the user.
• If authentication is successful, Phone will use the IP address
assigned by PPPoE server.
• After setting or getting correct IP Address, you can enter WebJeeves.
• Start Internet Explorer (IE6 with SP2, Service Pack) of PC
connected to the Ethernet Switch. Type in the IP Address
programmed above in the Address field. On the login page, enter
default password ‘1234’.
• To configure the SIP Parameters, proceed as mentioned below:
• Click on “SIP Parameters” from the menu provided on left side of
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the web page.
Get Information from your ITSP for SIP Account
•
•
•
•
•
•
•
SIP ID
Registrar Server Address
Registrar Server Port
Authentication User ID
Authentication Password
Outbound proxy Server address
Outbound proxy Server Port
• Enter above information in corresponding fields. If these
information is not available, keep the values as default. For e.g.
Registrar Server Port = 5060.
• Select ‘enable’ for Status-field for SIP1 and also enable Outbound
Proxy, if information is programmed.
• If you have a second internet phone Service Account (SIP2), then
use above fields for SIP2 port and enable.
• Submit page.
Note:
The PC need not be dedicated to the SETU VP236 after installation
is complete.
• You can have a look at all above parameters and software version
information by clicking on ‘Status’ from the menu. This page will
show following parameters which the SETU VP236 is currently
using:
• IP Address
• Subnet Mask
• Gateway Address
• DNS Address
• MAC Address
• SETU VP236 is ready for further IP calling service!
• User can also use Auto Configuration feature if he wants to avail
facility of configuration of SETU VP236 by ITSP from central
location.
• Default the system and start again, if you cannot establish
connectivity from your SETU VP236.
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Making Phone Calls
• After SIP1/SIP2 indication on the LCD, user can Go OFF-Hook, get
dial tone and press digits, wait for ‘Inter Digit Wait Timer” and the
number will be dialed out, if keypad is not locked. If it is
locked, the Phone will ask ‘User Password”. On entering correct
password, phone is unlocked.
• User can initiate a call by three ways as mentioned below:
• Using Handset
• Using Speaker
• Using Headset
• Hot Pad dialing
• User can use any method of dialing but after making a call, he
can switch between handset, headset and speaker mode.
• User can also make a call by using following features:
• Using the Phone Book
• Using the Speed Dial
• Using Call Logs
• Using a selective SIP trunk
• For Example, you can press ‘SIP2’ key and dial a SIP ID number
like [email protected]. For this follow the logic for
entering alphanumeric characters.
• You can also dial SIP ID number like ‘4567’ for making proxy call.
Receiving a Call
• When you get Ringing indication on Phone, go OFF-Hook and talk
by any of the ways mentioned below:
• Using the Handset-Pick up the Handset Æ Start talking.
• Using the Headset-Press the Headset key Æ Start talking.
• If Auto-Answer enabled-Without pressing headset key, the call
will be answered after some time (programmable) and will get
connected to the remote user, after going to Headset Mode.
• Using the Speaker-Press the Speaker key Æ Start talking.
• Pressing the Call Appearance key.
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Section 2: Features and Facilities
Matrix
3-Party Conference
Introduction
• Matrix supports the advanced feature of the phone, ‘3-Party
conference’. With this feature you can:
• Conduct a 3-Party Conference.
• Terminate a 3-Party Conference.
• Toggle between two remote users.
• Disconnect one of the parties and come to 2-Party Speech.
• Disconnect your self and connect remote user1 and remote
user2.
• Call Hold is not supported during active conference.
• Conference begins when you press the ‘Conference’ key. The LED
of the key will glow.
Conference is possible:
• From Menu
• Using a key, assigned
How to use it?
Conducting a 3-Party Conference using ‘Conference’ Key:
• To initiate a conference, press the Conference key while in a
conversation.
• After the user presses the Conference key, remote user1, will be
placed on hold and user will get a new dial tone.
• The user can now dial the 2nd party’s telephone number.
• When 2nd party answers the call, the conference will be
established.
• When the user hangs-up, all parties will be disconnected.
Terminating a 3-Party Conference:
To terminate Conference, the user will disconnect by going ON-Hook.
All other parties are released.
• Toggle between two remote users:
• The user will press the CA key for the remote user who has to be
held. Pressing the CA key holds the respective remote user
whereas the other remote user remains in speech with the
remote user. Thus remote users can be put in speech alternately.
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• Disconnect one of the parties and come to 2-Party speech:
• Pressing Conference key during active conference prompts/offers
the user a Menu. The user can select disconnection of remote
user1 or remote user2 (This feature can be used for multiple
conference).
• Disconnect himself and connect remote user1 and remote
user2:
• While in Conference, the user can press ‘Transfer’ key. This will
put remote users in speech and user himself will be
disconnected.
Configuring:
Conference is possible:
• From Menu. Refer the topic ‘Menu’.
• Using a key, assigned. For this click on Web Page, “Key
Programming”.
• Assign a key with Single Colour LED for this feature.
Important Points:
• Each conference call is logged as an independent call.
• If the user performs Call Mute, both the remote users will not be
able to listen user’s voice.
• Waiting call can be attended in the same way as in case of a
normal call. Refer chapter ‘Call Waiting’.
• Conference can be performed even for Peer-to-Peer
Configurations.
Relevant Topics:
1. Call Hold 56
2. Call Waiting 72
3. Call Appearances 49
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Anonymous Call Rejection
Introduction
• Some Incoming calls are from some unknown party whose Name/
Number is not displayed on the Phone. This type of calls without
CLI numbers or Name (Identity) is called Anonymous calls. Matrix
SETU VP236 supports a feature to reject such calls.
• Using this feature user can allow such calls as well.
• Configurable using ‘Web Page’ and through ‘Config Phone’ Menu.
How it works?
• This feature works for IC calls only.
• ‘Anonymous Calls Rejection’ can be set/canceled through Web
Jeeves or SETU VP236 (also through ‘Config Phone’).
• If set, the anonymous calls will be rejected.
• The rejected calls due to this feature are stored in the ‘Rejected
Calls’ List, along with the calls rejected due to other reasons like
manual call rejection or due to DND, etc.
Configuring:
Anonymous calls rejection-enable/disable can be configured through
Web or Phone.
• To configure ‘Anonymous calls rejection’ through Web-Refer the
topic ‘Web Jeeves’ for more details. Click on Web Page “User
Settings”.
• To configure from phone, access Config Phone-Menu and select
set/cancel.
Important Point:
• Anonymous call rejection is given priority over Call Forward. If
feature is set the anonymous call gets rejected before checking for
any Call Forward status.
Relevant Topics:
1. User Settings 152
2. Know Your SETU VP236 17
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Auto Answer
Introduction
• When the user is busy with important work then for any IC calls, the
speech of calling party can be routed to the Headset Port by the
phone. This type of feature is called ‘Auto Answer’.
• Incoming calls will be connected, after the time period of ‘Auto
Answer Timer’. For example, in ‘Call Centers’ persons receive
many calls. It is not practically possible to press a button or lift the
handset for each incoming call. In such cases, this feature is
useful using Headset.
• Auto Answer can be set/cancel:
• From Web Jeeves
• From Menu
• Using a key assign
• Key assigned should be with single color LED for this feature.
How it works?
• This feature works only when Phone is in Idle state.
• Auto-Answer status and timer are stored even when phone is
powered OFF.
• Caller will be connected to the user after time of approx. “AutoAnswer Timer”, which is programmable.
• The user will get a beep of 500ms On, 500ms Off and then both
the parties are connected.
• The key to set/cancel the feature works in toggle mode.
• Pressing the key will set/cancel Auto Answer. When set, the LED is
glowing on the key. When cancelled, the LED is turned OFF.
• If key for Auto-Answer is not assigned, the status of Auto-Answer
whether Set/Cancel can be known from the Menu. When the AutoAnswer option is selected, the display (while scrolling menu), will
show the status of Auto-Answer whether set or cancel.
Configuring:
• Auto-Answer can be set/cancel:
• From Web Jeeves
• From Menu
• Using a key
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• Auto Answer Timer is programmable. Range of Auto-Answer Timer
is from 00 to 99 seconds. Programming the Auto-Answer Timer can
be done through Web Jeeves and Phone Configuration. Program
‘00’ for immediate answer.
• Refer the topic ‘Web Jeeves’. Click on Web Page ‘System
Parameters’.
• To program Auto-Answer Timer from Phone access Menu to set/
cancel the feature.
• A key can be assigned for Auto-Answer set/cancel feature. Refer
Web page, “Keys Programming”.
Important Point:
• You should ware Headphone if auto answer feature is set.
Relevant Topics:
1. Receiving a Call 119
2. Menu 100
3. Key Assignment 87
4. System Parameters 144
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Auto Call Back
Introduction
• The user would not like to dial the busy station repeatedly. He
wants that if the called user is busy then his call should be queued
and should be made when the called user becomes free after
receiving notification within specified period (Notified Timer). This
feature is called ‘Auto Call Back’ (ACB).
• The feature can be configured by using the ‘Key’ assigned and
notification timer is programmable by Web Page.
• The key assigned is only with single colour LED (Do not assign key
number 06 to 09).
How to use it?
To Set/Cancel ACB:
• If the called party is busy, press the key assigned for ACB or select
an option in the Menu. The assigned key works in toggle mode.
How it works?
• When user is getting busy tone /busy message, he can set ACB by
pressing the ACB key or selecting an option in the Menu. When
ACB is set, message will be displayed on LCD for 2 seconds and
the LED on the key will be turned ‘ON’.
• When the called party is free, it will send ‘ready-for-call completion’
notification (ACB Notification) and the user gets the ‘Ring’
indication for this. Now he can make a call by pressing the ACB
key. LED on the key will be tuned OFF and ACB character on LCD
will be cleared when the call is made. This is like a normal OG call
made using SIP1, SIP2 etc.
• If another ACB is set during busy state before the first one getting
served, the earlier ACB request will be canceled and new ACB will
be applicable.
• User can cancel ACB by selecting an option in the Menu or
pressing the ACB key before receipt of the notification (ring).When
ACB is cancelled, the message will be displayed for 2 seconds and
LED will be turned Off.
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ACB Notification Timer:
• If ACB notification is NOT received within this time:
• ACB request for the particular number will be cleared
• LED will be turned off.
• ACB character will not be displayed on the LCD.
• If ACB notification is received within this time:
• User will get Ring if he is On-Hook (or when he goes ON-hook)
and will get dial tone if he goes OFF-Hook during ringing.
• ACB LED will flicker 500msec ON 500msec OFF.
• ACB character will flicker 500ms ON 500ms OFF on the LCD.
Configuring:
• Using phone, the feature can be set/cancel from ‘Menu’. For key
assignment, refer topic, Key Assignment.
• “ACB Notification Timer” can be programmed by clicking on Web
page “System Parameters”.
Important Points:
ACB Status:
• When the ACB option is selected from ‘Menu’, the display will show
the status of ACB whether set or cancel with a cursor on the
relevant row.
ACB in Power down mode:
• ACB status and ACB timer will be stored while Power down. The
timer will continue from where it had stopped.
• ACB request will not be stored while Phone is powered down.
Hence the user should again ‘set’ the required feature of ‘ACB’
when the phone is powered up.
ACB subscription from the remote peer (calling party):
• The called party will get ACB subscription request from the remote
peer only when no CA is free.
• If at least one CA is idle on receipt of ACB subscription, the ACB
notification will be sent to the calling party.
• Maximum four callers can subscribe for ACB.
Relevant Topics:
1. Key Assignment 87
2. System Parameters 144
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Auto Configuration
Introduction
• SETU VP236 configuration includes some basic settings like
Registrar Server Address, Authentication user ID, and User
Password which are provided by the Service Provider (ITSP). Also
some special services like Call Forward are provided by the ITSP.
Hence, it is desirable that the SETU VP236 is configurable by the
ITSP from the central location. Matrix SETU VP236 supports this
by providing Auto Configuration feature.
• The ITSP will keep the folders of configuration files in Central
Server for different SETU VP236 units as per MAC Address
number of the SETU VP236. When the customer connects the
SETU VP236 to the ITSP’s network, the SETU VP236 gets
automatically configured and the customer starts using the
services.
How it Works?
• SETU VP236 makes use of TFTP (Trivial file Transfer Protocol)
server to down load the configuration files.
• User has to just enter the server address provided by ITSP and
specify the config folder path provided by ITSP, in the Web page
for ‘Auto Configuration”. Then after selecting option for using the
feature at power ON, Click on “submit” the page.
• Three options for using the feature are provided:
• User can select that the SETU VP236 is not to be configured.
• User can select that the SETU VP236 is to be configured,
whenever it is switched ON.
• User can select that the SETU VP236 is to be configured, when
switched ON next time.
• SETU VP236 can use the address of TFTP server provided by the
DHCP or static IP Address as programmed by ITSP.
• If the FTP server address is not found, the auto-configuration
process is aborted and the current files are used by the SETU
VP236.
• If all the required files are not available in the folder, then there will
be incomplete configuration download to the SETU VP236. Also
only those downloaded files whose version will match to the
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version of SETU VP236 will be updated.
• For example, 7 configuration files are necessary and if only 5 files
are present, then these 5 files only will be downloaded by the IP
Phone and will be updated in SETU VP236 which may not meet
the required configuration of the SETU VP236.
• When Auto Configuration is complete, the ‘Status’ page displays
following information:
• In ‘Status’ field suitable message is displayed such as:
“No Tftp Server Found”
“Auto Configuration Partial”
“Auto Configuration Success”
• In ‘Statistics’ field, suitable message is provided regarding how
many files are down loaded successfully and total number of
configuration files updated by the system. For example, if
X=Number of files down loaded and Y=Total number of files
updated. Then the message displayed will be as:
“Files Downloaded/Configured = X/Y”
How to program?
Click on ‘Auto Configuration’ from the menu on the ‘Welcome Page’.
Auto Configuration:
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Matrix
Parameter
Value/Default
Description
At Next Power Default = Selected. Select this option if you want to
Auto Configure your IP Phone
ON
when you power it ON next time.
At Power ON
Not Selected
Never
Not Selected
TFTP Server
DHCP
Static
Default = DHCP
Select whether you will use
TFTP Server address provided
by DHCP or Static provided
by ITSP.
TFTP Sever
Address
N.N.N.N,
40 characters Max.
N = 001 To 255
Or Domain Name
Default = Blank .
40 Characters max.
Default = Blank.
Enter here the server address
provided by ITSP. Do not enter if
it is to be allocated by DHCP
server, on request by IP Phone.
Config Folder
Path
Select this option if you want to
Auto Configure your IP Phone
whenever you power it ON.
Select this option if you do not
want to Auto Configure your IP
Phone.
Provided by ITSP. Enter here the
path of the folder in which the
configuration files are stored.
For e.g. Matrix/SETU VP236.
Important Point:
• SETU VP236 will not restart, if user clicks on ‘Submit” on this page.
Relevant Topics:
1. Status 138
2. Software Upgrade 134
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Black Listed Caller’s List
Introduction
• Using this feature it will be possible to reject incoming calls from
specific addresses or contacts.
• List is configurable using ‘Web Page’ and using phone.
How to use it?
• The caller’s number which is to be rejected is programmed in black
listed caller’s list.
How it works?
• This feature is applicable for IC calls only.
• System supports a list with maximum 10 addresses (10 Entries).
• The number of calling party will be matched upto the last character
with the entries in the list.
• The rejected calls due to this feature is stored in the ‘Rejected
Calls’ list along with the calls rejected due to other reasons like
manual call rejection or due to DND, etc.
• If DND is set for “DND List” and calling party address is present in
both “DND List” and “Black Caller List”, the system will not obey the
DND List and will reject the call.
Configuring:
• Numbers can be entered in the list by following ways:
• Refer Web Jeeves and click on “User Settings’. For more details
refer chapter ‘User Settings’. By default, the list is Blank.
• The list can be configured from the Phone:
• Navigating Menu > Config Phone > Black Listed Caller List.
Important Points:
• Anonymous call gets rejected before checking for black callers list.
• Black listed caller is given priority over Call Forward. The call gets
rejected before checking for Call Forward status.
Relevant Topics:
1. User Settings 152
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2. Configuring the Phone-Menu 75
3. Anonymous Call Rejection 39
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Call Appearances
Introduction
• Matrix Phone supports four calls which are separate or joined. That
means if during an ongoing call a second incoming call comes
there will be an indication about that call and user can put the first
call on Hold. Thus, there are two active calls which user can select.
These two calls can be indicated on the LCD display. Now if third
and fourth calls will coming then last two calls will also be indicated
by the phone as ‘waiting calls’, Thus, we can have indications for
‘four’ calls maximum. This is known as Call Appearance (CA) for
the SETU VP236.
• Matrix SETU VP236 supports maximum of ‘4’ Call Appearances by
default. This is programmable.
• Similarly for OG calls also CA is applicable.
• The state of the call will be indicated on the corresponding key for
the CA.
How it works?
Selecting CA:
• The user can attend any call by switching between these four calls.
Assigning a Key:
• Indication on the Key-LED is updated after answering the CA.
• Keys can be assigned for each Call Appearance. If keys are not
assigned for Call Appearance then features like answering a call
selectively, answering a waiting call, toggling between two calls,
rejecting a call is not possible.
• By default, Keys 06-09 are used for Call Appearances. These
are dual color keys.
Key Number Feature
09
Call Appearance 1
08
Call Appearance 2
07
Call Appearance 3
06
Call Appearance 4
Refer the topic ‘Key Assignment’ for more details.
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Programming Number of CA:
• Number of Call Appearances can be programmed. For example, if
the user programs say, 2 then normal display will show the
information of only first two CA.
• If user has configured only two CA and gets a third IC call then in
such a case, it will be automatically rejected by the SETU VP236
so that user is not prompted to select that call.
• One Call Appearance is occupied for each IC call received/OG call
made. This will happen in increasing order viz. CA1, CA2, CA3
followed by CA4.
For Example:
Four CA can be used for four simultaneous calls.
• Consider a case for detailed explanation:
• Suppose that user is busy with two calls: First one is an OG call
occupying CA1 and second one is an IC call occupying CA2. The
user is talking over CA1, whereas CA2 is held.
• Now, if the user tries to make third call, then current call will go on
hold and user will get dial tone.
• If the user gets another IC call, it is treated as waiting call and is
indicated by CA4.
Call Toggle:
• Suppose there are two calls, one active and the other held. The
user can toggle between these two callers just by pressing a key.
Refer the chapter ‘Call Toggle’.
• Since we support four CA’s, four calls can be handled
simultaneously.
LED Indication:
• The status of the call (Call state) is indicated on the LEDs
embedded in the keys assigned to the CA feature. Refer ‘Getting
Started’ for more details.
How to use it?
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•
•
Click on Web Page “Key Programming”. Refer chapter ‘Key
Assignment’ for more details.
Program the keys for CA1, CA2, CA3 and CA4. Default is
06 to 09 for CA4 to CA1 respectively.
Important Points:
• It is preferable to program the key for “Rejecting a Call” as this will
be useful to reject the CA if required.
• Do not press the CA-Key and another key simultaneously. The
SETU VP236 can read only a single key being pressed at a time.
Related Topics:
1. Call Waiting 72
2. Making a Call 94
3. Receiving a Call 119
4. Key Assignment 87
5. Menu 100
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Call Forward
Introduction
• Matrix SETU VP236 supports many supplementary services
features like a normal Phone. For example: Call Hold, Call Forward
(CF), Call Transfer, etc.
• The phone supports Call Forward-Busy, Call Forward-No Reply
and Call Forward-All.
• Call Forward-Busy: Incoming calls on a user can be forwarded to
another designated party if the user is busy.
• Call Forward-No Reply: Incoming calls on a user can be
forwarded to another designated party when there is no response
from the user.
• User can forward his calls to one destination number-when busy
and to another destination number-when no reply.
• Separate key can be provided for Forward-Busy and Forward-No
Reply, feature for the phone.
• User can get the ‘status’ of Call Forward whether set or cancel.
•
Call Forward can be set/cancel:
• From Menu
• From Web
• Using a key.
• Keys with dual color LED should NOT be assigned for Call
Forward. Key with single colour should be assigned.
How to use it?
To set Call Forward:
• The user can press the pre-assigned Call Forward key (Busy or No
Reply).
• Navigate the menu using Up/Down keys. Select the option ‘Set’.
Enter the destination number to which the calls are to be
forwarded. Press enter key after entering the number.
• LED on the key will be ON when Call Forward is set.
To cancel Call Forward:
• The user can press the preassigned Call Forward key (Busy or No
Reply).
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• Navigate the menu using Up/Down keys. Select option ‘Cancel’
press enter key.
• LED on the key will be turned ‘OFF’ when feature is ‘Cancel’.
Call Forward Timer:
• This timer is applicable for Call Forward-No Reply only.
• All the calls are forwarded to programmed destination number after
time period of ‘Call Forward Timer’.
• ‘Call Forward Timer’ can be programmed from configuring phone
menu and Web page, also.
• You can set Call Forward-All, by selecting the Call Forward Timer to
‘0’.
Fox Example:
• If destination number programmed for Call Forward-No Reply is
7662 (SIP ID) and timer is 30 seconds, then system will wait upto
30 seconds. If the phone doesn’t reply, then the IC call will be
forwarded to 7662.
Configuring:
• Call Forward can be configured:
• From Menu (refer chapter-’Menu’)
• From Web page
• Using a key (refer chapter Key Assignment)
• After pressing the key, you will get a Menu for set/cancel. Select
‘set’ to configure the feature.
• Call Forward Timer can be programmed by:
• Configuring the Phone Menu (refer chapter)
• Web Page
• Click on Web page “User Setting” for programming through web
page.
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Parameter
Call
Forwardwhen Busy
Call
Forwardwhen No
Reply
Value/Default
Description
Select if you want to use the
Select/Blank
feature.
Enter here the destination
Default = Blank or
number to which you want to
ASCII characters
forward the IC call. If your
max. 40
phone is busy.
Select if you want to use the
Select/Blank
feature.
Enter here the destination
Default = Blank or
number to which you want to
ASCII characters
forward the IC call. If you
max. 40
cannot go OFF-Hook.
• Click on Web page, “System Parameters”. Refer chapter ‘System
Parameters’.
Parameter
Value/Default Description
Call Forward 00-99 seconds, Enter the timer value after which
Default = 45
you want the Phone to forward he
Timer
seconds
number. If programmed call
forward timer is 00 sec and call
forward-no reply is set, and then
all the incoming calls will be
forwarded to desired destination
number immediately.
Important Points:
• Call Forward set/cancel status and the destination addresses are
retained during power down, of the phone.
• Call Forward-No Reply is applicable for waiting call also.
• Black Caller list is given priority over Call Forward. The call gets
rejected before checking for Call Forward, if the number is in the
list.
• Call Forward is given priority over DND.
• Each forwarded call is logged, in the system.
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Relevant Topics:
1. Do Not Disturb (DND) 83
2. System Parameters 144
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Call Hold
Introduction
• The user can support multiple calls by using the ‘Hold’ feature. This
feature allows you to keep a call on hold and retrieve the call when
required. It also allows making second call/answer a waiting call
while holding a call.
• This feature is useful for various requirements for using the phone
with multiple calls, as explained below:
• User can ‘Hold’ Maximum of four calls and can toggle between
calls as per requirement. If number of CAs are 4, four calls can be
made/received simultaneously.
• User can retrieve the held call.
• User can make second call after holding a call.
• User can receive a waiting call after holding the current call.
• User can disconnect a held call.
• User can be reminded of held call.
• The held call will be disconnected, if not retrieved before
programmed time.
• ‘Hold’ can be accessed by navigating the features of ‘Menu’ or by
assigning the feature to a key using Web page ‘Key
Programming’.
• Select the key with No LED.
How to use it?
• To hold a remote user during speech, press ‘Hold’ key or
press the CA key of the active call.
• To retrieve the held remote user, press respective CA key.
• Call Hold and Retrieval can be done from ‘Menu’ also.
How it works?
Holding a Call
• It is possible to hold and retrieve a call when the phone is in
speech (connected)
• In all other states, call hold request is ignored.
• Held call with Headset or Speaker:
• Phone will go to idle state.
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• The Speaker/Headset will go silent
• The CA key blinks as per the cadence mentioned in the topic ‘Call
Appearances’.
• The LCD display will display the caller ID and the text ‘Hold’.
• Held call with Handset:
• User will get silence
• The user will go ON-Hook
• Phone will be Idle
• Indication of Held Call:
• The user gets silence.
• He is ready to do any other activity.
• If the call is through handset, the user may go ON-Hook.
• Held call is displayed on the LCD.
Retrieving Held Call
• User gets information about call retrieval by following ways:
• User gets connected to the held party.
• Held call will be changed to active call and hence the LCD display
will display the call as a normal call.
• If the call is retrieved while in speech with other remote user, the
current remote user will be held.
Alert for ‘Hold’ and Timer:
• ‘Call Hold Alert Timer’ is not separate for each call held.
• User will not get any hold alert if the phone is off-hook.
• If user goes on-hook with at least one held call, ‘Call Hold Alert
Timer’ will be started. Before the expiry of ‘Call Hold Alert Timer’, if
user goes off-hook, he will get dial tone of next free CA. But if no
CA is idle, he will get error tone.
• If user doesn’t go off-hook till the expiry of ‘Call Hold Alert Timer’,
he will get a ring to up to the time period of Ring Timer and all the
held calls are disconnected.
• User will be connected with the held call on lowest CA, if there are
multiple held calls and user goes off-hook before the expiry of ring
timer.
• Matrix SETU VP236 has a facility to Alert the user by following
ways:
• The Caller ID will flash for 500ms ON, 500ms OFF.
• LED of the respective CA will Glow/Blink. Refer the topic ‘Call
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Appearances’ for LED blinking cadence and color.
Rejecting Held Call and Timer:
• Held call rejection is indicated to the user in following ways:
• The row which occupied the held call is cleaned and the contents
are shifted to any of next rows upwards.
• Turning OFF the LED of the respective CA. Refer the topic ‘Call
Appearances’ for LED blinking cadence and color.
Configuring:
• ‘Call Hold Alert Timer’ is programmable only through Web Jeeves.
Click on Web Page ‘Timers’ for this. Refer chapter ‘Timers’.
• Range of ‘Call Hold Alert Timer’ = 000 to 999 seconds.
• Assigning 000 second to Call Hold Alert timer will mean that these
timers are not applicable. Refer Web pages for more details.
Remote Held:
When remote user puts you on ‘Hold’ it is called ‘Remote Held call’.
During this you can not ‘Hold’ that remote user. If you go ON-Hook,
when on same CA of remote held call, the call will be disconnected.
• But during Remote held call you can:
• Make a new call by-Pressing respective CA key.
• Receive IC Call by-Pressing respective CA key.
• Disconnect the call if you go ON-Hook with one Remote Held call.
Indication of Remote Held Call:
• You can get indication of remote held call by:
• Getting beeps.
• Getting Display on LCD.
• Indication of blinking of Respective CA LED.
Note:
• If there are multiple remote held calls and user goes OFF-Hook
before the expiry of ring timer, the user will be connected with the
remote held call on lowest CA.
• Remote held Call will be Rejected if user presses ‘Reject’ key.
Resuming Remote Held Call:
• In case the remote user resumes the ‘Held’ call, you can get
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indication when you are in ON-Hook condition by:
• Getting Continuous Ring
• Display of flashing ‘Rhold’ of respective CA.
• Indication of blinking respective CA LED.
• In case the remote user resumes the ‘Held’ call, you can get
indication when you are in OFF-Hook condition by:
• Getting beeps. like tone. This is called ‘Remote Held Resuming
Tone’. It is same as CCWT.
• Display of flashing ‘Rhold’ of respective CA.
• Indication of blinking respective CA LED.
Retrieving Held Call:
• In case the remote user retrieves the ‘Held’ call, you can get
indication by:
• Getting connected to the remote party and the Remote held call
will be changed to ‘Normal’ call and you will get ‘Display’ on LCD
for this.
Important Points:
• The user can answer the held call by pressing the respective CA
key. This might lead to Call Toggle. Please refer ‘Call Toggle’ for
more details.
• This feature is related with other features as mentioned below:
• Call Transfer: A held call can be transferred.
• Peer-to-Peer Calling : P2P can also be held.
• Rejecting a Call
: It is possible to reject a held call.
Relevant Topics:
1. Call Toggle 68
2. Call Appearances 49
3. Call Waiting 72
4. Receiving a Call 119
5. Making a Call 94
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Call Logs
Introduction
• This feature is provided to know the history of calls. With this
facility you can view:
• Last 20 dialed calls.
• Last 20 answered calls.
• Last 20 unanswered calls.
• Last 20 rejected calls.
• Last 20 forwarded calls.
• It is also useful to dial the number after viewing such numbers from
the Call Logs menu. The log will be as shown at end of chapter.
• This feature can be assigned a key with No LED. Feature can be
viewed on the Web Page.
How to use it?
• To view the log, press the key for ‘Call Logs’ or select ‘Call Logs’
option from the menu to view all the dialed numbers, select the
option ‘Dialed’. To view all the calls which are answered, select the
option ‘Answered’ etc.
• Using the phone you can scoll the ‘Menu’, and select ‘Call Log’
after selecting the type press ‘Enter’ key to see call details. Use <
key to go to previous menu. Finally to dial the selected number, go
OFF-Hook or press ‘Speaker’ key or press the ‘Headset’ key.
How it works?
• The call details can be viewed in brief or in details from the ‘Menu’
or using a key.
• Feature works during all call states of the phone while receiving, or
dialing etc.
• System will retain the Call Log even during power failure also.
• Call log contains following parameters for a call:
• Number
• Name of the called party
• Details of the call about Date and Time
• Duration of the call
• On pressing ‘Call Logs’ key or selecting the Call Logs option from
the ‘Menu’, a list of options for the call type is available:
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• Dialed
• Answered
• Unanswered
• Rejected
• Forwarded
• For any option, the details of Name, Number and Date and Time
whichever is available, is displayed on the LCD.
• If Name or Number (in case of calls other than dialed ones)
is not available for a call, it will be displayed as ‘ANONYMOUS’.
Dialed Number:
• All the dialed numbers, for making calls are stored.
Answered Number
• The Number is considered as an ‘Answered’ number when the user
answers the number of the received call. The Number is stored at
the end of the call.
Unanswered Number
• The Number is considered as an ‘Unanswered’ number when the
user does not answer it till ring time is expired. The number is
stored at the end of Ring Time Out Timer.
Rejected Number
• The number is considered as a ‘Rejected’ number when it is
rejected by the user manually. The numbers rejected due to lack of
CA or DND or Black Listed Caller’s logic are also stored as
‘Rejected’ number.
Forwarded Number
• The number is considered as a ‘Forwarded’ number when it is
forwarded by the phone as per the configuration setting of the
Phone. This number is stored after receiving the call forward
notification.
Dialing a Number
• Go OFF-Hook/press ‘Speaker’ key/press ‘Headset’ key from dialed
numbers’ list (when displayed in brief or in detail) to make an OG
call or second call.
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Configuring:
• User can assign a key for this feature. Refer web page ‘Keys
Programming’.
• ‘Call Log’ feature can also be selected from ‘Menu’.
How to use Web Page?
Click on Web Page, “Call Logs” to view the log.
• The parameters of the Log for each call are as shown below:
• Name: Name of the calling party. If name is not available,
‘Anonymous’ is seen in place of name. If call is not anonymous, it
is ‘blank’.
• Number: Here calling party’s number is displayed. If call is
anonymous, it is blank.
• Date: Date of the call is displayed in DD-MM-YYYY format.
• Time: Time of call is displayed as HH-MM, in 24 hour format. For
example, call at afternoon 3.00 hours will be displayed as 15:00.
• Duration: It is a total duration of the call, starting from the time of
call maturity, in the format: HH:MM:SS.
• You can select the type of call for which log is required. For
example, if you click on “dialed”, the phone will display the log of
all ‘dialed numbers’. Similarly you can click on: Answered,
Unanswered, Forwarded and Rejected calls.
• Clear all: Click on this to delete all the entries in call logs of
selected type. If selected log is for ‘dialed’, you will find the alert
message “This will delete all dialed entries”. Similarly, alert
message is found for other types.
Important Point:
• All the calls of the ‘Conference Call’ are stored as independent
calls.
Relevant Topics:
1. Menu 100
2. Know Your SETU VP236 17
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Call Mute
Introduction
• While you are in speech, during active call, you can hear the
second party but second party can not hear you. This feature is
called “Call Mute”.
• Call Mute can be set/cancel:
• From Menu
• Using a key
• The key to be assigned is with single color LED.
How to use it?
• To Mute the Call:
• Press the Call Mute key if assigned or use ‘Call Mute’ option
from the Menu.
• Press the Mute key/select ‘Set Call Mute’ option from Menu
• The call will mute and LCD will display ‘Call Mute Set’.
• The remote user will not be able to listen you.
• Press the Mute key again/selecting ‘Cancel Call Mute’ option
from Menu, call will unmute and LCD will display ‘Call Mute
canceled’. Now the remote user will be able to listen you.
How it works?
• This feature is applicable only for an active current call.
• When a call ends, Call Mute is canceled and not applicable for next
call.
• When Call Mute is set:
• OG speech is disconnected and the remote user will hear silence
(and not noise).
• LCD will display a text message for 2 seconds and come back to
the previous state.
• When Call Mute is canceled:
• OG speech is connected back and the remote user can hear the
user.
• LCD should display a text message for 2 seconds and come back
to the previous state.
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Configuring:
• For key assignment, refer web page ‘Keys Programming’.
Relevant Point:
1. Key Assignment 87
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Call Progress Tone Generation (CPTG)
Introduction
• Different types of tones are played during specific event of
establishing the call, speech duration and termination of the call.
This is called Call Progress Tone Generation (CPTG). As per ITU
standard these tones are of different specification for different
country. For example, Dial Tone for India consists of 400Hz
modulated by 25Hz, but for USA/Canada, it is 350+440Hz, without
modulation. Based on specific frequency, modulating frequency
and cadence, tones are categorized as shown below:
• Dial Tone
• Ring Back Tone
• Busy Tone
• Error Tone
• CO Call Waiting Tone (CCWT)
• Matrix SETU VP236 supports CPTG for many countries.
• SETU VP236 plays the Dial Tone when it is ready to accept the first
digit of a remote address to make an outgoing call. Normally
played when FXS goes OFF-Hook.
• SETU VP236 plays the Ring Back Tone when the called device is
alerting the callee.
• SETU VP236 plays the Busy Tone when the callee is busy.
• SETU VP236 plays ‘Error Tone’ when, there is no registration with
proxy server or user has performed some wrong operation.
• SETU VP236 plays the ‘CCWT’ only for first incoming call (Waiting
Call) which arrives while the user is connected to another party.
How it works?
• SETU VP236 supports call progress tones for various countries.
• Frequency and Cadence of “Call Progress Tone” for a country is
fixed (not programmable). Refer the table at the end of the chapter
for details of the frequency and cadence for different CPTG for
countries.
How to program?
• Click on Web Page, “System Parameters” Refer Web Jeeves.
• Refer chapter “System Parameters’ for CPTG programming.
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• Select Country: Click on combo box and select the ‘Country Name’
from the list of following countries:
Country
Dial Tone
Freq.
Cadence
Hz
second
Australia
425*25
cont.
Argentina
425
cont.
Ring Back Tone
Freq.
Cadence
Hz
second
0.4on 0.2off
400*25
0.4on 2.0off
425
1.0on 4.0 off
Brazil
425
cont.
425
1.0on 4.0 off
China
450
cont
450
1.0on 4.0off
Egypt
France
425*50
440
cont
cont
425*50
440
2.0on 1.0off
1.5on 3.5off
Germany
425
cont
425
1.0on 4.0off
Greece
425
0.2on 0.3off
0.7on 0.8off
425
1.0on 4.0off
400*25
cont
425
425
400
425
400
cont.
cont
cont
cont.
cont
India
Indonesia
Iran
Israel
Italy
Japan
Kenya
425
cont
Korea
350+440
cont
Malaysia
425
cont
Mexico
425
cont.
New Zealand
400
cont.
Philippines
Poland
Portugal
Russia
425
425
425
425
cont.
cont.
cont.
cont.
425
cont.
425
425
cont.
425*24
South Africa
400*33
cont.
400*33
Spain
Thailand
425
400*50
cont.
cont.
425
400
0.4on 0.2off
0.4on 2.0off
0.4on 0.2off
0.4on 2.0off
1.5on 3.0off
1.0on 4.0off
450
2.0on 4.0off
450
cont.
350+440
cont.
UK
350+440
cont.
USA/Canada 350+440
cont.
0.3on 0.3off
Error Tone
Cadence
second
425
0.375on 0.375off
425
0.3on 0.4off
425
0.25on 0.25 off
450
0.7on 0.7off
450
440
0.5on 0.5off
0.25on 0.25off
425
0.24on 0.24off
425
0.15on 0.15off
0.4on 0.2off
0.75on
400
400
0.25on 0.25off
0.4on 2.0off
0.75off
425
1.0on 4.0off
425
0.5on 0.5off
425
0.25on 0.25off
425
1.0on 4.0off
425
0.5on 0.5off
425
0.25on 0.25off
400
1.0on 3.0off
400
0.5on 0.5off
400
0.25on 0.25off
425
1.0on 4.0off
425
0.5on 0.5off
425
0.2on 0.2off
400*20 1.0on 2.0off
400
0.5on 0.5off
400
0.25on 0.25off
0.67on 3.0off
0.2on 0.6off
425
425
425
0.2on 0.6off
1.5on 5.0off
0.2on 0.6off
440+480 1.0on 2.0off 480+620 0.5on 0.5off 480+620 0.3on 0.2off
0.4on 0.2off
425
425
0.5on 0.5off
425
0.5on 0.25off
0.4on 2.0off
0.25on
425
1.0on 4.0off
425
425
0.25on 0.25off
0.25off
0.4on 0.2off
400+450
400
0.5on 0.5off
400
0.25on 0.25off
0.4on 2.0off
425+480 1.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off
425
1.0on 4.0off
425
0.5on 0.5off
425
0.5on 0.5off
425
1.0on 5.0off
425
0.5on 0.5off
450
0.33on 1.0off
425
0.8on 3.2off
425
0.4on 0.4off
425
0.25on 0.25off
Saudi Arabia
UAE
425
Freq.
Hz
400*25
Singapore
Turkey
Busy Tone
Freq.
Cadence
Hz
second
0.375on
425
0.375off
425
0.3on 0.2off
0.25on
425
0.25off
0.35 on
450
0.36off
425*50 1.0on 4.0off
440
0.5on 0.5off
0.48on
425
0.48off
1.2on 4.6off
0.4on 0.2off
0.4on 2.0off
0.4on 0.2off
400+450
0.4on 2.0off
440+480 2.0on 4.0off
400+450
425
0.5on 0.5off
425
0.25on 0.25off
425
0.75on
0.75off
425
0.25on 0.25off
400
0.5on 0.5off
400
0.25on 0.25off
425
400
0.2on 0.2off
0.5on 0.5off
425
400
0.25on 0.25off
0.3on 0.3off
0.2on 0.2off
450
0.5on 0.5off
450
0.6on 0.2off
0.375on
0.4on 0.35off
400
400
0.375off
0.225on 0.525off
0.375on
0.4on 0.35off
400
400
0.375off
0.225on 0.525off
480+620 0.5on 0.5off 480+620 0.25on 0.25off
CCWT
Freq. Cadence
Hz
second
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400
400
400
400
400
400
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400
400
400
400
400
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
400 0.3on 6.0off
By default, Country is India.
Remarks:
Frequency notations are as:
• f1*f2: f1 is modulated by f2.
• f1+f2: The juxtaposition of two frequencies f1 and f2 without
modulation.
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Important Point:
• Rx volume of Rx Speech tones can be increased/descreased by
using left and right arrow key on the phone.
Relevant Topic:
1. Volume Settings 160
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Call Toggle
Introduction
• It is often required for the user to toggle between an active on
going call and a held call.
• For this Matrix SETU VP236 provides the feature called ‘Call
Toggle’.
• There is no extra key assignment for this feature.
How it works?
• This feature is a just an extension of multiple CA.
• While answering/retrieving a call, the active call will get held and
the user will get connected to the remote party.
For Example:
•
•
•
Suppose CA1 has an active call whereas, CA2 has a held call.
Now if you wish to talk to held call on CA2, just press the key for CA2.
This will act as retrieving a call since CA2 has a held call.
The current call i.e. on CA1 will get held and CA2 will be active.
• User can toggle between two calls during call conditions: Ringing
or Matured.
• User will not be able to toggle between calls in any other states.
• Whenever a user is held while toggling, all the actions applicable
for a held call like Call Hold Alert Timer, etc. are applicable. For this
refer chapter ‘Call Hold’.
Relevant Topics:
1. Call Appearances 49
2. Making a Call 94
3. Making Second Call 99
4. Receiving a Call 119
5. Call Hold 56
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Call Transfer
Introduction
• As three parties are involved for this feature, following terms are
used to explain the feature:
• Transfer Target: The remote user to whom the call is to be
transferred. (1st party).
• Transferor: The user who transfers the call.
• Transferee: The user who is to be transferred (2nd party)
• Call Transfer is possible:
• From Menu
• Using a key (without LED)
• “Transfer Notification Timer” to provide information about ‘Transfer’
to the user is programmable from Web page “System Parameters”.
• Matrix SETU VP236 supports two types of Call Transfer:
• Unattended (Blind) Transfer
• Attended Transfer
Unattended or Blind Transfer
• In this type the user can transfer the active call to the transfer
target irrespective of his state of calling. User need not first talk to
him before transferring.
• The user will get notification for the status of transfer.
Attended Transfer
• For this type of transfer, user will first talk to the transfer target
party before transferring the call without the transferee party being
able to listen.
• While talking to transfer target, if the user presses ‘Transfer’ key,
call will be transferred, but CA2 will not be released. The port will
be idle.
• The CA1 and CA2 will display the transferee and transfer target
number till the expiry of transfer notification timer OR 5 seconds
after successful notification ‘Conc’ received from transferee.
• ‘Xfer1’ will be displayed on CA1 and ‘Xfer2’ will be displayed on
CA2 for the cadence: 500ms ON 500ms OFF.
• The system will wait for the notification from the transferee. If
notification is not received within transfer notification time, both the
CAs will be released.
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How to use it?
Unattended or Blind Transfer
• To initiate an unattended or blind transfer:
• Press the ‘Transfer’ key while in a conversation.
• The 1st call will be placed on hold and you will get a new dial
tone.
• Dial the 2nd party’s telephone number.
• The call will be transferred after Inter Digit Wait Timer time.
• You will be notified by Remote user1 about the transfer action i.e.
failure/successful provided remote user1 supports notification.
Attended Transfer:
• To initiate attended transfer:
• Press the ‘Hold’ key while in a conversation.
• The 1st call will be placed on hold and the user will get a new dial
tone, when selecting second CA after pressing CA key.
• You can now dial the 2nd party’s telephone number. When the
2nd party answers, you can have a private conversation with the
2nd party without the 1st party being able to listen.
• If the 2nd party wants to speak with the 1st party. Press Transfer
key to complete the call transfer.
• You will drop out of the conversation with both parties, but the call
will not be disconnected, and you will get suitable notification.
Transfer Notification:
After transferring the call, the user will get information or notification
about the status of transfer. This will be displayed for a period of some
time known as “Transfer Notification Timer”.
• User will get display of the transfer target’s number on the CA on
which transfer activity is done.
• The display will be for the time of Transfer Notification Timer.
• If notification is not received within transfer notification time, CA is
released.
• Call transfer notification (Display) (Blind Call Transfer)
• Transfer target busy: ‘Busy’.
• Transfer target ringing: ‘Free’.
• Transfer target ringing over; ‘Disc.’
• Transfer target answered: ‘Conc.’
• Transfer target error: ‘Error’. Now if the user goes ON-Hook in
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this state, the user will get ‘Ring’ indication. When the user goes
OFF-Hook, remote user1 is retrieved and connected to the user.
• Call Transfer Notification (Attended Call Transfer)
• Call transfer notification is provided for following cases of transfer
target :
• Transfer target busy
• Display will be ‘Busy’. The CA2 (transfer target) will be
released immediately after notification is received and CA1
will be released after the expiry of transfer notification time.
• Transfer target answered.
• Display will be ‘Conc’. The CA1 will be released after 5
seconds and CA2 will be released immediately after ‘Conc’
notification is received.
• Transfer target error.
• Display will be ‘Error’ till transfer notification timer.
• Now if the user goes ON-Hook in this state, user will get Ring.
When the user goes OFF-Hook, the transferee will be retrieved
and will be connected to the user (transferor). The CA1 will be
released after the expiry of notification time and the CA2 will be
released immediately after ‘Error’ notification is received.
Configuring:
This is a feature that does not need any configuring except assigning
a key for it.
• Call Transfer can be done:
• From Menu
• Using a key
• If key for Call Transfer is not assigned, the user can select ‘Call
Transfer’ from Menu.
• Key with no LED can be assigned for this feature. For key
assignment, refer topic, ‘Key Assignment’.
• “Transfer Notification Timer” is programmable from Web Page
“System Parameters”.
Relevant Topics:
1. Call Hold 56
2. Key Assignment 87
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Call Waiting
Introduction
• Normally, the user will come across many calls. Some calls may be
coming to him at the same time when he is busy with an active
call. The Phone supports to handle these calls by putting them in
the ‘Waiting’ state.
• Matrix SETU VP236 supports the IC call as Waiting Call with
following conditions. For all the condition, if user accepts the
waiting call, the active call will be held.
• The user has one active call.
• The user has one held and one active call.
• The user has one held, one active and one waiting call.
• The new IC call received with one held, one active and two waiting
call is not considered as waiting call and will be rejected.
How it works?
• A waiting call can be answered by pressing a respective CA keys
• After answering a waiting call the active call will be held.
• Number of waiting calls depends on the number of CA
programmed.
• The user can answer the waiting call by pressing the respective CA
key. This will be like Call Toggle. Please refer ‘Call Toggle’ for more
details.
Relevant Topics:
1. Call Hold 56
2. Call Appearances 49
3. Call Toggle 68
4. Call Progress Tone Generation 65
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Change Password
Introduction
• The SETU VP236 configuration is user specific. Hence user should
be able to select his personnel password to replace the default
password.
• User can change both types of password from its default value:
• Configuring Password: Used to enter the ‘Configuring the Phone’
menu from the main Menu, and also to enter ‘Web Jeeves’.
Default = 1234.
• User Password: Used to unlock keypad. Default = 1111.
Configuring:
• Click on Web ‘Change Password’.
How to use Web Page?
Parameters
Value/Default Description
Configuring Password
Enter New
4-digits
Enter the new password. If
Password
maximum 0-9 you want to change the
Password. Remember it.
Retype
-doEnter same password as
Confirm
above
Password
User Password
Enter New
4-digits
Enter the new password.
Password
maximum 0-9 Remember it.
Retype to
-doEnter same password as
Confirm
above
Password
Relevant Topics:
1. Menu 100
2. Keypad Lock 88
3. Web Jeeves 168
4. Default SETU VP236 79
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CLIR
Introduction
• CLIR is “Calling Line Identity Restriction”. This feature is used by
the user when he doesn’t want to disclose his identity to the
called party.
• Assign the key for CLIR with single color LED.
How to use it?
• Set/cancel the feature from Web page ‘User Setting’.
• Press the key assigned to set the CLIR. LED will glow.
• By using phone, you can navigate Menu to select CLIR and press
‘Enter’ key.
How it works?
• The key assigned for CLIR set/cancel feature will work in toggle
mode.
• When ‘set’, LED of the key will glow. So that user is notified that
CLIR is set. When ‘cancel’, the LED of the key is turned OFF.
• If key for CLIR is not assigned, and when the CLIR option is
selected from the Menu, the display will show the status of CLIR.
• The phone will retain the CLIR status while power down.
Configuring:
• For key assignment, Clock on ‘Keys Programming’.
• Refer the topic ‘Menu’ to know how to set/cancel CLIR from Phone
(Menu).
• Click on ‘User Setting’ to set/cancel from Web. Refer chapter
‘User Settings’.
Relevant Topics:
1. Menu 100
2. Know Your SETU VP236 17
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Configuring the Phone-Menu
For more details, refer chapter “Know Your SETU VP236”.
How to use it?
• Select the parameter, ‘Config Phone’ from scrolling the ‘Menu’
Parameters.
• Press “Enter” key. Enter User Password.
• Navigate ‘Config Phone Menu’ by up/down key.
For entering “Config Phone Menu”, follow the steps mentioned below:
• Press ‘Menu” key, if assigned or select ‘Menu’ from LCD display
and press ‘Enter’ key.
• Navigate the list with Up/Down key and select “Configuring the
Phone” option.
• Press ‘Enter’ key. System will ask ‘Password’.
• On entering correct password (default 1234), you can navigate the
menu using up/down key.
• Select the feature by scrolling the list and press ‘Enter’ key.
For example:
• User can select ‘DND List’ to edit the number in the List, displayed
on the LCD Display.
• All the five editable keys are used for editing.
• Here, < key will not work as ‘back key, rather it will shift the cursor
on the LCD display to the left by one character.
• After editing when the user presses ‘Enter’ key, the number/
address is saved whereas the cursor comes to the initial position.
• The user may use Up/Down key to change the rows.
• To exit the mode, user should press < key (back key). Whatever
changes have been made will be saved.
Related Topics:
1. Know Your SETU VP236 17
2. Menu 100
3. Do Not Disturb 83
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Daylight Saving Time
Introduction
• It is the new local time assigned to a region for a portion of the
year, usually an hour forward from its standard official time.
• The clocks are advanced at the beginning of the daylight saving
time and delayed at the end of the daylight saving time.
• DST can be set according to day-month wise or date-month wise.
• DST can be disabled by the SE if not applicable in the country.
• DST is changed on a particular day and month every year (say
Type1) for some countries. The DST is changed on exact date and
month (say Type 2) every year for some other countries and
remaining countries don’t follow DST.
• If DST type 1 is selected, the DST is effective on the programmed
day and the month whereas, if DST type 2 is selected, the DST is
effective on the programmed date and the month of the year.
Configuring:
• Click on Web page “Daylight Saving Time Adjustments”.
Program the parameters as per the DST rules of the Country where
the IP Phone is installed.
Forward Time Adjustments:
Type: Select the type from
• None
• Day-Month wise: Program this option only for a country where DST
is observed as per Day-Month wise every year.
• Date-Month wise: Program this option only for a country where DST
is observed as per Date-Month wise every year.
Default = None.
Day-Month Wise:
• On: Program ‘day’ and ‘month’ on which DST will be applied every
year. For this, program Ordinal, Day and Month. For this option,
‘Type’ should be selected as ‘Day-Month wise’. Default = 1st.
Sunday January.
Ordinal: Program 1st, 2nd, 3rd, 4th, or 5th. For example if DST starts on
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first Sunday of January every year, then program ‘1st’ here.
Note:
• For last Sunday (or other day) of the month, always set Ordinal
Number = 5th. For e.g. If the month has 4 Sundays in a particular
Calendar year then the last Sunday will be automatically the fourth
one and if the month has 5 Sundays in a particular month of the
Calendar year then the last Sunday will be automatically the fifth
one, and since while programming it will not be known that in a
given year, the last Sunday will be fourth or fifth one, user should
program 5th for last day of month.
Day:
• Program the day on which DST starts. Select from Sunday to
Monday of the week. In above example, select ‘Sunday’.
Month:
• Program the month on which DST starts every year. Select the
month from January to December of the year. In above example,
select ‘January’.
Change Time From:
• It is the current time settings at which DST will start to change.
The time is in HH:MM (Hours-Minutes) format, where HH=00 to 23
and MM=00 to 59. Default = 00:00.
To:
• Enter forwarded time here at which DST will be forwarded. Enter
the time in HH:MM (Hours-Minutes). Default = 00:00.
Date-Month Wise:
• On: Program ‘date’ and ‘month’ on which DST will be applied every
year. For this, program Date and Month. For this option for ‘Type’
should be selected as ‘Date-Month wise’. Default = 01 January.
• Date: Program the date on which DST starts. Select from ‘01’ to
‘31’ of the month. For example if first January is the DST date, then
select ‘01’
• Month: Program the month on which DST starts every year. Select
the month from January to December. In above example, select
‘January’.
• Change Time From: It is the current time settings at which DST
will start to change. The time is in HH:MM (Hours-Minutes) format.
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Default = 00:00.
• To: Enter forward time here at which DST will be forwarded. Enter
the time in HH:MM (Hours-Minutes). Default = 00:00.
Backward Time Adjustments:
• Program all the parameters as explained for ‘Forward Time
Adjustment’ except the ‘To’ time. Default values are same.
• To: Enter the backward (delayed) time here at which DST will be
backwarded (delayed).
Note:
• If for some country, the Backward Time Adjustments is at 00:00
hours, use previous date with “from” time = 23:59 and “to” time as
required.
Example 1:
• In New Zealand, the DST starts on Last Sunday of October. The
clock changes from 02:00 to 03:00. Hence, parameter ‘On’ will be
as 5th Sunday, October as per Day-Month wise option, of ‘Forward
Time Adjustment’.
Example 2:
• In New Zealand, the DST ends on Third Sunday of March. The
clock changes from 03:00 to 02:00. Hence, parameter ‘On’ to affect
DST in New Zealand, will be programmed as 3rd Sunday, March as
per Day-Month wise option of ‘Backward Time Adjustment’.
Example 3:
• In Cuba, the DST starts on 1st April of every year. The clock
changes from 01:00 to 02:00. Hence, parameter ‘On’ will be
programmed as ‘01 April’ as per Date-Month wise of ‘Forward Time
Adjustment’.
Example 4:
• In Cuba, the DST ends on last Sunday of October every year. The
clock changes from 23:59 (in fact 00:00 midnight) to 23:00. Hence,
parameter ‘On’ will be programmed as 5th Sunday, October as per
option ‘Day Month wise’ of ‘Backward Time Adjustment’.
Relevant Topic:
1. Time Settings 148
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Default SETU VP236
Introduction
• Some times while user is configuring and programming the Phone,
he would like to start from beginning and wants to reset the Phone
to its factory set values.
• This is done by the feature “Default the SETU VP236” Matrix
phone can be defaulted only from the Web Page. But the
passwords can be default using phone as explained below.
Default Password using keypad:
• The SETU VP236 supports two passwords: Config Password and
User Password.
• If you default the system, these passwords will also get default.
But if you forget the config password, you can not enter ‘Jeeves’
to change the password. To avoid such problems, the SETU
VP236 supports following procedure:
• Go OFF-Hook. You will get ‘dialtone’. Navigate ‘Menu’ to select
‘Alphanumeric Dialing’.
• Press characters ‘reset’ the string will be dialed out.
• You will get dial tone after resetting of both passwords and a
message will be displayed for reseting of password.
How to use it?
• Click on web page ‘Default IP Phone’.
• Click on ‘Yes’ when the phone shows Alert message.
• Using phone, navigate Menu > Alphanumeric Dialing > Press
charaters ‘reset’.
• You will get dial tone after reset.
For Example:
• ‘This shall assign default values to all the parameters of SETU
VP236, do you want to continue?’
• Click on ‘OK’ if you want the SETU VP236 to get settings as per set
by factory.
• Click on ‘Cancel’ if you want to continue with current status of the
Phone.
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Relevant Topic:
1. Change Password 73
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Dial Plan
Introduction
• When user has two SIP accounts, he can use SIP1 for making calls
to US and SIP2 for making calls to Gulf. He may like to have a
facility to select the SIP Line by the system automatically for a
specific Number to be dialed out. Matrix provides such facility by
providing a feature called “Dial Plan”.
• This feature is programmable by Web Page “OG Call Route
Selection”.
How to use it?
• Program the number to be dialed in the dial plan table.
• Dial that number.
• Call will be made using ‘SIP Account’ as per the table.
How it works?
• The system will make an OG call based on the Number Dialed.
• The Dial Plan table has maximum 10 entries.
• Each entry has two parameters viz. Address and SIP line.
• The Number to be entered is of maximum 40 characters.
• SIP line can be SIP1 or SIP2 account as only two lines are
supported by the Phone.
• Now Whenever an OG call is made, the number to be dialed out on
SIP is compared with the entries in this table. If a match is found
the number is dialed using the respective SIP account, else the
number is always dialed using OG call Route selection logic.
Example 1:
If the user wishes to make all calls starting with 33 through SIP1 but
calls made to numbers starting with 331 through SIP2, he should
enter 33 in first index with SIP1 and 331 in the second index with
SIP2.
Example 2:
Suppose that the dial plan table is programmed as shown below:
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Index
01
02
:
10
Address
5678
56
:
SIP Account
SIP1
SIP2
:
If user dials say 567894, the call is made using SIP1 as this number
best fits with the number in index 01.
But if the user dials 561234, the call is made using SIP2 as this
number best fits with the number in index 02.
How to program?
Click on “OG Call Route Selection”.
Refer corresponding chapter for more details.
Related Points:
1. OG Call Route Selection 110
2. Selective SIP Line 127
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Do Not Disturb (DND)
Introduction
• Many times, user needs privacy for some duration and do not want
to receive any calls without logging off from the server or switch
OFF the phone. But at the same time, he would like to receive calls
from specific callers. For such requirements, Matrix SETU VP236
supports a feature “Do Not Disturb” (DND).
• User can set/cancel DND.
• User can set DND for all callers or for few callers and allow other
callers to make calls on his address.
• User can use ‘Menu’, key or Web page for configuring this feature.
The key assigned is with single color LED.
How to use it?
• Set DND, by pressing the DND key assigned
• Navigate the menu using Up/Down keys and select the desirable
option and press ‘Enter’ key.
How it works?
• DND is applicable for IC calls only. OG calls can be made as usual.
• User can allow maximum 10 callers to call him even if he has set
DND. The numbers of such callers can be programmed in a list
called ‘DND List’.
• DND list can be programmed from Web Jeeves as well as from
Phone (Menu Æ Configuring the Phone).
• DND can be set/canceled:
• From Menu
• Using a key
• From Web
• When DND key is pressed user will get following three options:
• DND All
• DND List
• DND Cancel
DND All:
• If the user selects DND All, no incoming call will be accepted. The
call will be rejected by the SETU VP236 with cause ‘Busy’. LED
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assigned, will glow.
DND List:
• If the user selects DND List, the caller ID of the IC call is fully
matched with numbers in the DND List and if the number matches
the call will be accepted else it will be rejected with cause ‘busy’.
Refer chaper ‘Configuring the Phone-Menu’ for programming the
list.
DND Cancel:
• If user selects ‘DND Cancel’ then ‘DND’ feature is not applicable
and the SETU VP236 will work normal as per other settings.
Configuring:
DND List can be configured through Web or Phone.
• DND through Web-Refer the topic ‘Web Jeeves’ for more details.
Click on the Web page ‘User Setting’.
• DND through Phone-Navigate Menu > DND.
Important Points:
• Black Caller list is given priority over DND. The call gets rejected
before checking for DND.
• Each rejected call due to DND will be logged as rejected call.
Rejected calls log may also contain the calls rejected manually by
the user or calls rejected due to Black Caller List.
Relevant Topics:
1. Call Forward 52
2. Rejecting a Call 122
3. Configuring the Phone-Menu
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Hotline
Introduction
• In order to avoid entering and dialing of long digits every time, user
will prefer that these numbers are dialed out by the system,
automatically.
• Matrix SETU VP236 provides this feature by allowing
preprogrammed number to be dialed by the SETU VP236 after a
short delay. This feature is called “Hot Line’. The delay is
programmable, by parameter ‘Hotline Timer’.
• It can be programmed by Web page, Menu parameters and using
a specific key.
• Assign the key with single color LED indication, only.
How to use it?
• Set Hotline by pressing a key or using Menu options.
• Go OFF-Hook for the time of “Hot Line Timer”.
• The programmed number will be dialed out.
How it works?
• The status, Hotline address and timer will be retained when Phone
is powered OFF.
• If the user dials a number before expiry of the Hotline Timer, the
number dialed by the user is dialed.
• If Hotline is set but the hotline address is not programmed then
user will get error tone.
How to program?
• Hotline can be set/cancel:
• From Web
• From Menu
• Using a key (refer, ’Key Assignment’)
• Click on Web page “System Parameters” for configuring ‘Hotline
Timer’. Refer chapter ‘System Parameters’.
• Click on Web page, “User Setting” for programming ‘Hotline
Address’. Refer chapter ‘User Settings”. Configure Hotline feature
as shown below:
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Parameter
Hotline
Hotline
Address
Value/Default
Set or Cancel
Description
Select ‘Set’ if you want to
use the feature.
Default = Blank or Enter here the Address you
ASCII characters want to be dialed by the
max. 40
Phone when using the
Hotline feature.
Important Points:
• It is advisable for the user to set hotline timer less than First Digit
Wait Timer.
• If hotline timer set is more than First Digit Wait Timer, user will get
error tone after the expiry of First Digit Wait Timer and the
programmed hotline address will not be dialed.
Relevant Topics:
1. Menu 100
2. Key Assignment 87
3. System Parameters 144
4. User Settings 152
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Key Assignment
Introduction
• This chapter explains, how you can assign a specific key to a
specific feature.
• After assigning the key press that key to use the feature.
How to use Web Page for key assignment?
• Click on the Web Page “Keys Programming” for assigning the
specific number of key.
For Example:
Suppose you want to assign a key number 01 to a feature.
• Click on box for key number 1. The list appears. Select a feature
for which key number ‘01’ is to be assigned.
• User should assign the feature a single or dual color LED as per
requirement only.
BE CAREFUL TO SELECT:
A KEY WITH SINGLE COLOR LED or A KEY WITH DUAL COLOR
LED or A KEY WITH NO LED, FOR THE FEATURE WHICH IS TO
BE ASSIGNED. REFER RELATED CHAPTER FOR THE
REQUIRED KEY.
• Assign required feature to key number from 01-18. You can operate
the feature by pressing the ‘Key number’ assigned here. Refer
chapter ‘Know Your SETU VP236’ for list of feature.
• If you have selected “Phone Book” from the feature list, then select
the index number at which required contact phone number is to be
assigned as per the phone book table.
Relevant Topics:
1. Know Your SETU VP236 17
2. Phone Book 117
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Keypad Lock
Introduction
When the keypad is locked it is not possible to make OG calls.
Using this feature it is possible:
• To lock the keypad by pressing a key assigned or using the option
‘Lock Keypad’ from Menu.
• To lock the keypad automatically after a programmed time.
• To receive IC calls in locked condition by using CA keys, Headset
key, speaker key and Hook switch.
• To unlock the keypad the password is required.
• Assign the key with single color LED.
How to use it?
• To lock the keypad:
• Press the Keypad Lock key which is pre-assigned.
• Select the option ‘Lock Keypad’ from the Menu.
• To unlock the keypad:
• Press the keypad lock key again or press any other key, which is
assigned
• Enter the password (default 1111).
• If Auto keypad lock is enabled, the keypad will get locked after time
= Keypad Lock Timer.
How it works?
• This feature is applicable during any state of the Phone like
receiving, calling, etc.
• Auto keypad lock can be selected from Web as well as from Phone
(Config Phone Menu).
• This feature works with three parameters:
• Keypad Lock Methods: It has two modes viz. Manual and Auto.
• Keypad Lock Timer : It can be set from Web as well as from
Phone (Config Phone Menu).
• Keypad lock/unlock (manual) : It can done using a key/Menu.
•
Keypad lock (manual) : It can done using an option
‘Keypad Lock’ from user Menu.
•
Keypad unlock (manual) : It can done by pressing any
key and entering password.
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Keypad Lock : Manual
• You can assign a key for Keypad lock feature.
• On pressing the ‘Keypad Lock’ key when the keypad is open:
• The keypad is locked.
• LED of the key is ON.
• You will get display of a text message, ‘Keypad Locked’ for 2 Sec.
• Phone will go to idle state/standby state or previous state.
• Pressing the ‘Keypad Lock’ key or any other key when the keypad
is locked:
• System will ask ‘User Password’.
•
If you enter correct user password: (Default user p/w is 1111).
•
You will get display ‘Keypad Unlocked’ for about 2
seconds.
•
LED is turned OFF.
•
Phone goes to idle state/previous state.
•
If you enter incorrect user password:
•
You will get the message on LCD: ‘Invalid Password, try
again’ for about 2 seconds.
•
Go to idle/standby state or previous state.
•
If No password is entered:
•
You will get the message on LCD: ‘User Password not
entered’.
•
Phone will go to idle/standby state or previous state.
• Keypad lock status is stored while power down.
Keypad Lock : Auto
• When you make any activity/event, keypad lock timer is started.
• If no activity/event is done till expiry of keypad lock timer:
• Keypad is Locked.
• LED of the key is turned OFF.
• You will get text message as “Keypad Locked” for about 2
seconds.
• Phone goes to Idle/Previous state.
• Keypad status is retained even during power down.
Configuring:
• Auto Keypad Lock and ‘Keypad Lock Timer’ are configurable from
Web Jeeves and Phone (Config Phone Menu).
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• Click on Web Page, “System Parameters”. Refer chapter ‘System
Parameters’ to set/cancel ‘Auto Keypad Lock’ and ‘Keypad Lock
Timer’.
Important Points:
In Keypad Lock-Manual
• If the user forgets the password, he can change the user password
from Web Jeeves and open the keypad. Alternatively, the user may
default the SETU VP236.
• If key for Keypad lock is not assigned, the keypad can be locked
from the Menu.
In Keypad Lock-Auto
• If the call is being made and the keypad lock time gets expired then
the keypad will get locked. However, the call will not get
disconnected.
Relevant Topic:
1. System Parameters 144
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Last Number Redial
Introduction
• It is desirable that user should just press one key if he wants to dial
same number frequently. For such case a feature is required by
which a user is able to redial the last number by pressing a key or
selecting option from the Menu. This is called ‘Last Number Redial’.
• The key to be assigned will be without LED.
How to use it?
• Press the Last Number Redial key, which is preassigned.
• Last dialed number will get dialed out.
How it works?
• By pressing the assigned key for Redial, user can redial the last
number on which OG call was made or number dialed.
• To dial the last dialed number again, the last number can be
obtained from the last dialed numbers’ Call Log.
• If key for Redial is not assigned, the feature can be used from the
Menu.
• Calls with redialed numbers are stored in the Call Logs.
Configuring:
• Refer Web Jeeves and click on ‘Keys Programming’ for assigning
key number.
• To redial, using phone, select option ‘Last Number Redial’ by
navigating ‘Menu’.
Relevant Topics:
1. Know Your SETU VP236
2. Key Assignment 87
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LCD Settings
Introduction
• Matrix SETU VP236 supports the adjustment of the LCD contrast
and brightness to improve readability in various conditions. The
parameters like Backlight, brightness and Contrast can be adjusted.
• The Feature is configurable by Web page and “Config Phone”
Menu.
How it works?
Backlight:
• User can turn ON or OFF the Backlight of the LCD display on the
Phone.
• The Backlight will be turned ‘ON’ in any of the following conditions:
• User receives an incoming call.
• User goes OFF-Hook
• User dials any digit in any state
• User pressed any key in any state
• User goes ON-Hook
• When Remote end disconnects/ rejects/ held the call
• User receives any notification (ACB, Message Waiting
Notification, Transfer)
• User can also program the ‘time’ after which the backlight will be
turned OFF if no activity is performed on the Phone. For example, if
10 sec. is programmed, the Backlight will be turned OFF after 10
sec if no activity is done during this time.
• Thus following options can be programmed:
• Always ON
• Always OFF
• For 10sec
• For 30sec
• For 60sec
• For 120sec
• For 240sec
• By default it is ‘For 10sec.’
Contrast:
• LCD Contrast Level 1 to Level 4 can be set. Level1 will provide
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lowest and Level 4 will provide highest contrast.
Brightness:
• LCD Brightness Level 1 to Level 4 can be set Level1 will provide
lowest and Level 4 will provide highest brightness.
Configuration:
• Click on Web page “System Parameters”. Refer chapter ‘System
Parameters’.
• Refer topic ‘Config Phone Menu’ for more details, refer ‘Know Your
SETU VP236’.
Related Topic:
1. System Parameters 144
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Making a Call
Introduction
• The Matrix IP telephone provides advanced calling functionality. For
example, it can support multiple calls on a single line. It can also
support multiple lines.
• Maximum number of simultaneous outgoing calls is equal to the
number of CAs programmed. Means if number of CAs programmed
are four, maximum four calls can be made simultaneously.
• For Example, if number of CAs programmed are 3, first OG call will
be made from CA1 (if does not select explicitly other CA).
• If user selects CA2, first call on CA1 will go ‘held’ and user will get
dial tone on CA2 to make a new call.
• Similarly, user can make further calls (maximum four calls when
CAs programmed are 4) by holding the current call.
User can make a call by following methods:
• Using Handset
• Using Speaker
• Using Headset
• Hot Pad Dialing
These are explained below:
• Using Handset:
• Lift the handset, wait for the dial tone and then dial the desired
number.
• Using Speaker:
• Press the speaker key, Wait for the dial tone and then dial the
desired number.
• Using Headset:
• Press the “Headset” key, wait for the dial tone and then dial the
desired number.
• Hot Pad Dialing:
• This is a type of ON-Hook Dialing.
• Press the number directly being ON-Hook
• The speaker gets turned ON
• Dial the digits
• You will get call progress tones.
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• You can use any method of dialing but after making a call, you can
switch between handset, headset and speaker mode.
• User can also make a call by using following features:
• Using the Phone Book
• Using the Speed Dial
• Using Call Logs
• Using a selective SIP trunk
• Refer corresponding chapter to use these features.
How it works?
The system supports the feature “End of Dialing” so that system starts
dialing out the digits as per the way in which the digits are entered.
This is explained first followed by the methods for dialing the number:
• End of Dialing: SIP Number Dialing and IP Number Dialing.
• ON-Hook Dialing: Enter the number first and then dial the number,
also called Pre-dialing.
• OFF-Hook Dialing: Pick up the handset and then dial the number.
End of Dialing:
The Phone supports the feature by following options:
• You can press ‘Enter’ key, after dialing digits string.
• You can Program the Inter Digit Wait Timer after expiry of which
the system dials out the number.
• You can program the “Length’ of the “Fixed Number of Digits’. By
default, it is disabled.
• Program these in the Web page for “System Parameters”. You
can enable any of above option. If all are enabled then system will
dial out as per the first action detected. For example, ‘Enter’ flag is
enabled, number length is set to 5 and Inter Digit Timer is set to 6
seconds, then end of dialing is considered using ‘Enter’, if ‘“’ is
received after 3 digits. Likewise, end of dialing is interpreted if the
Inter Digit Wait Timer expires after few digits.
• By default, ‘Enter’ key is enabled, Inter Digit Wait Timer = 6
secs. and Fixed Number of digits Length = 40 characters. (For
more details refer chapter ‘Timers’).
Caution!
User should never program all options of “End of Dialing” as
“Disable”. Other wise system cannot dial any number.
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How SIP Number (SIP ID) can be dialed?
• Press a key which is assigned as ‘Alphanumeric Dialing Key’.
• This will indicate that SIP address is to be dialed.
• Dial the SIP ID. You can dial SIP ID which considers numbers first
and then alphabets. For example, if you dial 2, ‘2’ will be
interpreted first followed by a, b, c.
For Example:
You can dial a SIP ID number like [email protected] by first
pressing key which is assigned as ‘Alphanumeric Dialing Key’ and
then dialing the ID with all alphabets and numerical.
How IP number can be dialed?
• You can dial IP address by replacing ‘dot’ with ‘asterisk’
• You can dial using the same way as used for dialing a SIP ID. Both
the options can be used.
• If you dial the number using first option, the system will consider
the number to be an IP number when it receives ‘Asterisk’ as the
fourth character of the number string.
ON-Hook Dialing
• Phone is in Idle state.
• Dial the digits till End of Dialing.
• Number is dialed out as per Peer-to-Peer Table if best match is
there with number string in the Peer-to-Peer Table.
• If match not found, it will be dialed out using SIP account as per
OG call route selection logic.
• The speech is established through speaker. One Call Appearance
(CA) is occupied and the LED glows for the respective CA key.
The number dialed is displayed on the LCD in the CA selected. The
trunk through which the call is to be made depends on the OG
route selection method. If the call is made through SIP1 trunk, SIP1
is displayed in the call status characters. Likewise, ‘SIP2’ is
displayed, if the call is made through SIP2 trunk.
• If you dial end of dialing digit as the first digit, Error message is
displayed on the LCD.
OFF-Hook Dialing
• You can go OFF-Hook, pressing speaker key, headset key or lifting
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handset.
• Dial tone and state of First Digit Wait Timer.
• On expiry of ‘Timer’, you will get Error Tone for Error Tone Timer
and then phone will go in ‘Standby’ and ‘Silent’ state.
• You will get ‘error message’ on the LCD.
• Go ON-Hook. Phone will go in ‘Idle’ CA key is released, LED of
speaker key will be OFF.
• You will get ‘Idle’ message on the LCD.
• But if you dial any digit, during FDWT, and if you don’t dial any
digit till IDWT then dialed number string is dialed out call will be
processed further.
Examples for dialing different types of IP address/numbers:
• 12345/ vivek/vivek12345/*12345
• When you dial the SIP Address which contains only numerals/
alphabets/ combination of numerals & alphabets/ special
characters as prefix (‘*’ etc), then system will apply the OG call
routing logic and number will be sent to the SIP server found from
the OG routing logic.
• [email protected]
• If you dial the SIP Address which contains the IP address, the
Phone will compare the IP address dialed with the registrar server
address (any one among the two, as IP phone support two SIP
lines) programmed in SIP parameters configuration.
•
If the IP address dialed match perfectly with the registrar
server address programmed, then it will send the dialed SIP
address, to the same SIP server to which match found, using
the same SIP line.
•
If the IP address dialed doesn’t match with the registrar
server address programmed, then it will consider the call as
peer to peer call and the logic for peer to peer calls is used
for dialing.
• [email protected]
• If you dial the SIP Address which contains the domain name, the
Phone will compare the domain dialed by the user with the
registrar server address (anyone among the two as IP phone
support two sip lines) programmed in SIP parameters
configuration.
•
If the domain dialed match perfectly with the registrar server
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•
address programmed, then it will send the dialed SIP address
to the same SIP server to which the match found, using the
same SIP line.
If the domain dialed doesn’t match with the registrar server
address programmed, then it will consider the call as peer to
peer call and the logic for peer to peer calls is used for
dialing.
Important Points:
• If the user has two calls held by remote users, it will not be possible
to make new call since no resources are available to entertain the
new call. Refer chapter ‘Call Hold’ for ‘Remote Held’.
• However, user will get dial tone when selects the free CA to make a
new call. Since, no resources are available, call will not be
completed. But to make a call, user will have to release one of the
calls held by remote user.
• User can make a new call when user has Active Conference.
Maximum two simultaneous OG calls can be made when user has
active conference.
• When user has active conference which will occupy two CAs and
user selects the free CA to make new call, two calls involved in
conference will go ‘held’ and the user will get the dial tone.
Relevant Topics:
1. Switching between Handset, Headset and Speaker
2. Peer-to-Peer Calling 113
3. Call Hold 56
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Making Second Call
Introduction
• This feature allows you to make another call while holding current
call. It also allows disconnecting the second call and retrieving
back the first call.
• This feature is useful for using the phone with multiple calls, as
explained below:
• User can hold one call (IC or OG call) and make another call.
• User can toggle between the two calls. Refer topic ‘Call Toggle’
for more details.
• User can make 3-Party conference after establishing second call.
• User can transfer the held call to the second call.
How to use it?
You are busy with one active call, first call.
• You are in speaker mode. Press ‘Hold’ key.
• Dial the number string to make second call.
• Now CA2 is occupied.
The same logic can be extended for subsequent calls. For example,
maximum four simultaneous calls can be made if the number of Call
Appearances are programmed as 4.
Note:
• You can also select ‘Hold’ from the features by navigating the
‘Menu’, to make a second call.
Related Points:
1. Call Toggle 68
2. Call Appearances 49
3. Call Waiting 72
4. Call Transfer 69
5. Call Hold 56
6. 3-Party Conference 37
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Menu
Introduction
• Some features can be accessed from the Phone just by navigating
the ‘Menu’. It is a kind of ‘short cut’ method instead of entering Web
Jeeves, to set the feature.
• ‘Menu’ can be accessed by following ways:
• Using Normal Display on the LCD: Select the line displaying
‘Menu’ using up/down key and press ‘Enter’ key.
• Using the specific key assigned for ‘Menu’. The key assignment
can be done only from the Web Page, ‘Keys Programming’.
How it works?
• Menu key can be pressed and features can be navigated by up/
down key provided on the phone.
• For example, to configure parameters of SETU VP236 from the
phone itself, user will have to navigate the options of the ‘Menu’
and select the option “Configuring the Phone”. Note that SETU
VP236 will ask ‘Password’, if you want to enter the ‘Config the
Phone’. Similarly user can select “Hotline” to set hot line feature.
Following cases explain the use of Menu Mode:
If no activity during Menu mode:
• If the user does not perform any activity, he will go to Normal
Display after about 15 seconds or by pressing the Å key (Back
key).
OG/IC Call when not connected:
• You are in Menu state, then to make an OG call or receive an IC
call, you will have to come out of the Menu.
• Also In case of receipt of IC call, the phone will ring and the CA on
which the call has arrived will display normal IC call activities.
OG/IC Call when connected:
• When you are in connected state (during speech) and receive IC
call the call will be displayed on the LED of the CA, and you will get
indication by beeps for call waiting.
• In such case you can press the CA on which the IC call has
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arrived, to get connected to the arrived IC call.
Network Status:
• Following network parameter will be displayed on IP Phone if user
selects ‘Network Status’ from the menu:
• IP Address
• Subnet Mask
• Gateway Address
• DNS Address
• Domain Name
OFF-Hook during Search contact option:
• If you are in ‘Search Contact’ option of the Menu select the number
and go OFF-Hook, to make an OG call.
Accessing SIP Line:
• During ‘Menu’ mode if you use a feature like ‘Accessing SIP2’ while
talking to remote user1 then the current active call gets held and
you will get dial tone to dial out another number of remote user2.
Which features can be accessed?
Refer chapter ‘Know Your SETU VP236’.
Important Points:
• ‘Call Forward’ can be set by entering Destination Number from
keypad.
• When selecting ‘Config Phone’ Menu during speech, using keypad
digits will not affect current call or speech.
Related Topics:
1. Getting Started 27
2. Know Your SETU VP236
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Network Port
Introduction
• For considering variations of network conditions and usage by
various customers, Network Port for SETU VP236 are
programmable.
• For this Matrix SETU VP236 provides a feature called “Network
Port Parameters’.
• It can be programmed from Web page, but some parameters can
be programmed from the phone also.
• You can also get display of few network parameters by accessing
‘Network Status’ in ‘Menu’. Refer chapter ‘Menu’.
How it works?
Matrix SETU VP236 supports the network parameters:
• Configurable by Web
• Configurable by Phone
Configurable by Web
• Connection Type
• PPPoE User ID
• PPPoE Password
• Static IP Address
• Subnet Masks
• Gateway Address
• DNS Server
• DNS IP Address
• DNS Domain Name
• STUN Server Address
• STUN Server Port
• NAT Keep Alive Status-Enable/Disable
• NAT Keep Alive Interval
• SIP QoS Type
• SIP QoS Level
• RTP QoS Type
• RTP QoS Level
• Use VLAN
• VLAN Id
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• SIP Listening Port
• RTP Listening Port
Refer the chapters “Web Jeeves” and “VoIP Basics’ for detailed
explanation of IP Address, PPPoE, DHCP, STUN, NAT, etc.
Configurable by Phone, after selecting ‘Network Parameters’ from
Config Phone Menu.
• Connection Type-DHCP, PPPoE and Static
• IP Address
• Subnet Mask
• Gateway Address
• DNS Server-Automatic/Static
• DNS Server Address
• DNS Domain Name
• PPPoE User ID
• PPPoE User Password
• VLAN Setting
How to use?
• For programming by the phone, navigate Menu > Config Phone >
Network Parameter.
• Click on Web page “Network Parameters”. Refer Web Jeeves
for details of programming the parameters.
Connection Type:
Select IP Address allocation from three options:
• DHCP
: If clicked on DHCP, IP address, Subnet Mask, Gateway
Address will be replaced by the dynamic values sent by
the DHCP server.
• PPPoE : If clicked on PPPoE, IP Address will be allocated by
PPPoE Server.
• Static IP : If clicked on the option ‘Static IP’, you can use IP
Address by your ITSP. Default = Static IP.
By default, Connection type is ‘Static IP’.
Enter PPPoE parameters as shown:
• PPPoE User ID:
It is a string of maximum 16 characters. It can be extended ASCII
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character set.
By default, PPPoE User ID is Blank.
• PPPoE Password:
Enter a string of maximum 16 characters. It can be extended ASCII
character set.
By default, PPPoE Password is Blank.
Enter Static IP parameters as shown:
• IP Address:
It is relevant only if Connection Type = Static IP.
It is a string of maximum 15 characters.
(Enter the dot ‘.’ using the digit ‘1’ as used in case of alphanumeric
dialing).
Default IP Address is 192.168.001.161.
• Subnet Mask:
It is a string of maximum 15 characters.
By default, Subnet Mask is 255.255.255.0.
• Gateway Address:
Gateway Address is a string of maximum 15 characters.
By default, Gateway Address is Blank.
Make DNS (Domain Name Server) settings as:
• DNS Server:
• Obtain Automatically
• Static
If ‘Obtain Automatically’ is selected and connection type is DHCP,
then DNS address provided by the DHCP server, will be used. If
‘Obtain automatically’ is selected and connection type is PPPoE,
then DNS address provided by PPPoE server will be used. If static
is enabled then phone will use the DNS address programmed
manually. You cannot select ‘Obtain Automatically’ when connection
type is set ‘Static’. By default, DNS Server is static.
• DNS Address:
DNS IP Address is a string of maximum 15 characters using 0-9
digits. By default, DNS Address is Blank.
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• DNS Domain Name:
It is a string of maximum 24 characters. It can be extended ASCII
character set.
By default, DNS Domain Name is Blank.
Enter NAT Parameters as:
• NAT Keep Alive Status:
Select from Enable/Disable.
If enabled, the SIP messages will use NAT router’s public IP
address and SIP port number. The NAT will also be configured with
this port number to forward the traffic with this port number. NAT
eliminates the need for STUN.
By default, NAT Keep Alive Status is Disable.
• NAT Keep Alive Interval:
This timer specifies the time after which the SETU VP236 will send
specific SIP messages to the SIP server. Digits can be used from
001-200 Seconds.
By default, NAT Keep Alive Interval is 120 Seconds.
Enter STUN parameters as:
If NAT is configured do not program STUN:
• STUN Server Address:
STUN server address is the address of the STUN server. It can be
IP address also. Maximum 40 characters can be used. It can also
be extended ASCII characters.
By default, STUN Server Address is blank.
• STUN Server Port:
STUN Server Port specifies the STUN Server’s listening port for
SIP. User may change this field if the ITSP provides STUN server
port number other than the default.
Range is from 1024 to 65535.
By default, STUN Server Port is 3478.
Enter QoS (Layer3) parameters as:
• SIP QoS Type:
This field specifies the Quality of Serve (QOS) type. The SETU
VP236 sends all the SIP signaling messages with this setting.
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Select from two options:
• Precedence (also called TOS/Priority)
• DiffServ is used for voice traffic.
By default, DiffServ is selected.
• Enter the SIP QoS Level in the field provided.
‘Precedence’ uses 3 bits and hence has valid range from 0 to 7.
By default, Precedence is 5.
DiffServ uses 6-bits and hence has valid range from 00 to 63.
By default, DiffServ is 26.
• RTP QoS Type:
This field specifies the QOS type.
Precedence (Also called TOS/Priority).
DiffServ is used for voice traffic. The SETU VP236 sends all the
RTP media with this setting.
By default, DiffServ is selected.
• Enter the RTP QoS Level in the field provided.
Precedence uses 3 bits and hence has valid range from 0 to 7.
By default, Precedence is 5.
DiffServ uses 6-bits and hence has valid range from 00 to 63.
By default, DiffServ is 46.
VLAN Setting:
Use VLAN?
• Select the option ‘Yes’ from Yes/No if ‘VLAN Tag’ is required.
• If selected ‘Yes’, all the packets leaving the network port will include
the VLAN Id programmed otherwise the packets will be sent
untagged. Default, use VLAN option is No.
VLAN Id
• Program the Id also called ‘VLAN identifier’, if in above parameter
‘Use VLAN?’, ‘Yes’ is selected.
• Id can be programmed using Max. 4-digits from digits 0-9.
• Range = 0000-4095
• By default, VLAN Id is ‘0’.
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SIP Listening Port
• During SIP signaling, source and destination ports are specified for
the listening and transmission of SIP messages. This is specified
by ‘SIP Listening Port’. User can configure the SIP listening port as
per requirement.
• System will use common SIP listening port for all the IP calls.
Range = 1024 - 65535
By Default = SIP Listening Port is 5060.
RTP Listening Port
• RTP listening port is required to listen for RTP packets transmitted
by remote peer after RTP session being established. User can
configure the RTP listening port as per requirement.
• System will use unique RTP listening port for each RTP session
(for call on each CA).
• User can program only even port number (as per RFC standard) for
RTP listening port
Range = 1024-65526
By Default = RTP Listening Port is 8000.
Relevant Topics:
1. Configuring SETU VP236 16
2. Web Jeeves 168
3. Know Your SETU VP236 17
4. VoIP Basics 162
5. Peer-to-Peer Calling 113
6. VLAN Tagging 154
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Notifications
Introduction
• Matrix SETU VP236 has the facility to provide the indication to
the user that the Phone has responded the operation carried out by
him. This is done by providing suitable ‘Notification’, which will be
displayed on LCD for time of ‘Notification Display Timer’.
• Following features are supported for providing notifications:
• Call Transfer
• Voice Mail
• Call Forward
• ACB Notifications
• Remote Hold
For Example:
• Call Transfer Notification Timer: This timer signifies the time for
which the SETU VP236 should wait for the notification of transfer
function from the transferee. This timer is programmable.
• ACB Notification Timer: This timer signifies the timer for which
the SETU VP236 should wait for ACB ‘ready for call completion’
notification.
• ACB SET: If user has successfully set the Auto Call Back feature,
he will get this notification.
• For other features refer related chapters and web pages.
Notification Display Timer:
• This timer is the time for which the IP phone will display the
notification ‘Conc’ received from the transferee.
• Notification display timer is fixed of 5 seconds. The Call
appearance will be released after expiry of display timer.
• Notification display timer is applicable for both blind and attended
transfer.
• It is not used for other transfer notifications like ‘Ring’, ‘Busy’,
‘Error’ etc.
For Example:
• If transfer notification timer programmed is 30 seconds and
transferor receives the ‘Conc’ notification after 9 seconds after the
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transfer activity being performed, then the notification is displayed
for 5 seconds on the LCD and CA is released after 5 seconds.
Similarly for above case, if transferor receives notification ‘Conc’
after 27 seconds after transfer activity being performed, message
will be displayed on the LCD for 3 seconds only.
Important Point:
• Transfer Notification Timer will always have priority over notification
display timer.
Relevant Topics:
1. Call Transfer 69
2. Auto Call Back 42
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OG Call Route Selection
Introduction
• Matrix VP236 supports two SIP lines and different line can be
selected by phone while making an OG Call. This is called “OG
Call Route Selection”.
• This feature allows you to make a call with either of the SIP line
registered with ITSP.
• One important application of this feature is like a case in Least
Cost Routing (LCR). If you want to automatically select a low cost
SIP Trunk ‘SIP2’ whenever you make an OG call, select the SIP2
trunk on the Web page.
• SIP1 or SIP2 can be selected alternately if ‘Rotation’ option is
selected.
• This feature is programmable only through Web Jeeves.
How it works?
• “OG Call Route Selection” supports the following options:
• SIP1
• SIP2
• Rotation
• All the calls are made using SIP1 if SIP1 is programmed.
• All the calls are made using SIP2 if SIP2 is programmed.
• If ‘Rotation’ is selected, call can be made using SIP1 and SIP2
alternately.
• The Phone can select the SIP line automatically on the basis of
number dialed (Refer chapter “Dial Plan”).
Note:
• For ‘Rotation’ if last time calling had used SIP2-Line, then for
current dialing, phone will select the SIP1-Line.
How to use Web page?
Click on “OG Call Route Selection” from the menu.
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Parameter Value/Default Description
Call Route
Selection
Dial Plan
Number
SIP Line
SIP1, SIP2,
Rotation.
Default = SIP
1
Select the option for the ITSP
Registered SIP Account line on
which you want to initiate a call.
Max. 16 digit,
1-9, 0, #, *
Default =
Blank
Enter the Number which should
be dialed using specific SIPLine. Maximum 10-numbers can
be programmed with SIP Line
selected
Select the Registered SIP Trunk
line which will be selected by
the Phone while making OG
Call on the Number specified
SIP1, SIP2
Relevant Topics:
1. Dial Plan 81
2. SIP Parameters 128
3. Selective SIP Line 127
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PC Port
Introduction
• At this port you can connect the PC for configuring the VP236.
• This is one of the ports of the “Two Port Switch” which is inbuilt to
the VP236. The other port is Network Port, for connecting the
cable Modem or ADSL Modem. Hence, all the parameters of PC
Port are same as Network Port.
• Refer the chapter Network Port for detailed information.
Relevant Topic:
1. Network Port 102
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Peer-to-Peer Calling
Introduction
• Calling over IP network without using SIP server is called Peer-toPeer Calling (P2P Calling).
• You can call another SETU VP236 connected by the VLAN
Services provided by the ITSP, by the destination number which
can be a number with IP Address or Domain Name.
• This feature works by programming:
• Peer-to-Peer Config Table
• Peer-to-Peer Table
Mumbai Office
Delhi Office
Router
2001
Internet
Router
2002
192.167.1.21
192.168.1.15
IP Gateway
VLAN
Services
Peer-to-Peer Table:
• Suppose SETU VP236S are connected in Delhi office and Mumbai
office and these two sites in different cities are connected to each
other by the VLAN services.
• Then a call can be made from Delhi office to Mumbai office and
vice versa, without using SIP proxy server.
• Now there are two ways to reach each other: One way is to dial the
IP address. But many times, the offices use Gateways at both the
ends. In such case, another way is to reach a particular
station by dialing his station number followed by the IP address or
Domain Name, which is programmed as destination address.
Peer-to-Peer Configuration Table:
• User can use different codec’s or DTMF schemes to take care of
different Network conditions while making peer to peer call. Hence,
different configuration of codec’s and ‘dtmf’ option may be required
for peer-to-peer calls and proxy calls. These can be programmed in
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peer-to-peer configuration table.
• Program ‘Yes’ for the parameter, ‘Use STUN’. Use ‘Network
Parameters’ web page for programming STUN Server Address and
Port Number.
How to use it?
• Before entering Numbers and Destination Address for the peer-topeer call at different entries, configure the peer-to-peer
configuration table for these parameters;
• Local User’s Name
• Vocoder Preferences
• DTMF option
• ‘Use STUN’ option
• Prepare a table of your contact Numbers with their Destination
Address. Then program the information in the Web Page for ‘Peerto-Peer Table”.
How it works?
• “Peer-to-Peer Table” supports 500 entries.
• Each entry consists of the Number and corresponding Destination
address.
• The best fit logic is used to match the dialed string with
programmed destination address in “Peer-to-Peer Table”.
• If best match found, the call is considered as Peer-to-Peer call and
<dialed numbered>@Destination Address will be dialed.
• If best match is not found, the call is considered as Proxy call.
Suppose ‘Peer-to-Peer Table’ is programmed as follows:
Index
Number Dest. Address
001
12345
abc.com
002
123
192.168.1.37
:
:
:
500
• Now, if user dials say 123, the number dialed will be
[email protected], as it best match with the entry in Index 02.
• But if the user dials 12345567, the number dialed will be
[email protected] as this entry best fits with the number in
index 01.
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• For making an OG Call system will use the configuration for
sending Name, selecting Vocoder, DTMF option and STUN which
are programmed in ‘Peer-to-Peer Configuration Table’.
• User can use global (Unique) IP Address in case of WAN and
private IP Address in case of LAN to make P2P call.
How to use Web Page?
• Click on Web page, “Peer-to-Peer Configuration”. Refer Web
Jeeves for programming details.
• Click on Web ‘Peer-to-Peer Table’.
• Program both as explained below:
Peer-to-Peer Configuration Table:
Parameter
Display
Name
Value/Default
Max. 24-characers with
ASCII Default = Blank
DTMF option Default = SIP (INFO)
Vocoder
Preference
10-preferences. Refer
chapter SIP Parameters
for default.
Use STUN
Yes/No. Default = No.
Description
Enter here Local user’s Name
which you want to be displayed
at remote end.
Refer chapter SIP Parameters.
Phone will use the option
programmed here.
Select As per SIP Parameters.
Refer chapter SIP Parameters.
Phone will use the preference
programmed here.
It specifies whether STUN should
be used for P2P call or not. If
'Yes', STUN will be used to get
the IP Address. If No, the STUN
need not be used.
Peer-to-Peer Table:
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Parameter
Index
Value/Default
001 to 500
Description
Number at which entry is made.
001 to 100 can be programmed.
Click on links 101-200 or 201 to
300 or 301 to 400 and 401 to 500
to program other entries.
Number
Maximum8 digits, 0-9, *.
Default =Blank
Max. 40 chars. Including
ASCII characters.
Default=Blank.
Enter the extension number of
the user agent to be called
Enter IP Address or 'domain
name' of the user agent who will
be called by P2P Calling.
Destination
Address
Important Point:
Programming of Authentication ID/Password for SIP Account ‘1’:
• The Phone will use the Authentication ID and Authentication
Password of SIP Account1 when Peer-to-Peer call is made and
called party IP Phone (remote client) asks for Authentication. Hence
the call will be terminated if remote client challenged for
authentication but SIP Account1 is disabled or Authentication ID/
Password is not programmed.
Relevant Topics:
1. SIP Parameters 128
2. Making a Call 94
3. Network Port 102
4. Configuring the Phone-Menu 75
5. VLAN Tagging 154
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Phone Book
Introduction
• Storing of the Name and Number in the memory of the phone, is
very much useful to the user for using the SETU VP236. This is
called “Phone Book” feature. You can store maximum 100 contacts.
• You can program the number/name of a contact in the Phone Book
either from Web or through Phone (Using Configuring the Phone
Menu).
How to use it?
• Press the Search Contact key or select the Search Contact option
from the ‘Menu’.
• All the contacts, programmed in Phone Book is displayed in
alphabetical order.
• Navigate the Phone Book using up/down navigation keys.
• Press enter key on the name selected, to see the details (i.e.
number) of the contact.
• Press ‘Å’ navigation key to go back one step each time it is
pressed.
• Finally to dial the number, go OFF-Hook or press speaker key
or press headset key or press ‘Enter’ key.
How it works?
• In the Phone Book, 100 entries (contacts) can be entered with a
name and a number for each entry. By default, Blank.
• The numbers are stored index wise (as these are programmed
index wise) and not in alphabetical order in the Phone Book.
However, when the user uses Search Contact feature, he will get
the entries in alphabetical order.
• A contact can be searched from the Phone Book using the feature
‘Search Contact’.
Configuring:
• Configuring the Phone Book through Phone.
• Phone Book is one of the options in the feature ‘Configuring the
Phone Menu’.
• When the user selects the option Phone Book, a menu with
following options are available:
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•
Add new
•
Edit
•
Delete Specific Contact
•
Delete All Contacts
• The user may select any of the menu option using up/down
navigation key.
• The user may press Å navigation key to go to previous step.
• If the user selects an option ‘Add New’, phone will prompt the
user to enter Name followed by Number.
• How to use Web page?
• Please refer topic, ‘Web Jeeves’ for more details. Click on “Phone
Book”.
Parameter
Value/Default
Description
Number at which entry is made.
Index
01 to 100
Non programmable.
Enter here the Name of the
Name
24 digits max.
Default = Blank. Contact number whose SIP
number is to be stored in the
It can be ASCII
Phone Book.
Characters.
Number
40 digits max.
Enter the Number of the contact
Default = Blank. It party for whom the name is
can be ASCII
entered above.
Characters.
Important Point:
• While entering an IP address, ‘.’ i.e. dot is entered (dialing * is
interpreted as dot only while dialing and not while editing or
entering the number or address).
Relevant Topics:
1. Speed Dial 136
2. Know Your SETU VP236 17
3. Key Assignment 87
4. Search Contact 126
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Receiving a Call
Introduction
• Normally Incoming Calls to the phone will be on the first available
line of SIP1 or SIP2.
• The associated LED of the line-key (if assigned) will glow and the
phone will ring.
• If there are multiple lines ringing, you can select the line, which you
would like to answer.
• At the end of the call, hang up the handset to disconnect the call.
• Maximum calls the IP phone user can receive simultaneously is
equal to the number of CAs programmed. This means, if number of
CAs programmed are ‘4’, user can receive maximum of four calls
simultaneously.
• For Example, if number of CAs programmed are 4, first incoming
call will occupy the CA1.
• Second incoming (waiting) call will occupy the CA2. If user
answers the call on CA2, the current call will be put on hold and
user will come in speech on CA2.
• Third incoming (waiting) call will occupy the CA3. If user answers
the call on CA3, the current call (on CA2) will be put on hold and
user will come in speech on CA3.
• Fourth incoming (waiting) call will occupy the CA4. If user answers
the call on CA4, the current call will be put on hold and user will
come in speech on CA4. Now, user has four held calls and user
can toggle between the calls as per the requirement. All further IC
calls will be rejected.
Ways of answering a Call:
• Using the Handset
• Using the Headset
• Using the Speaker, If Auto Answer enabled
• Pressing the Call Appearance Key
Using the Handset:
• This is a most common method like analog telephones. You can
just pick up the handset before expiry of rings and start talking to
the caller.
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Using the Headset:
• In this method, you can press the Headset key and start talking.
If Auto-Answer enabled:
• This is a special feature, enabled on the Web page programming
or by the programming of key. After some time, call will be
answered automatically by the Phone and you will get audio of
calling party through head phone.
Using the Speaker:
• Like normal analog phone, you can press the speaker key and
start talking.
Pressing the Call Appearance key:
• When Incoming call comes and you are busy with an outgoing call,
the corresponding Call Appearance Key LED is glowing.
• Press the key and start talking.
• Refer chapter ‘Call Appearances’ for more information.
How it works?
How user gets information of the IC call ?
• When Phone is in Idle condition, ringing indication is available on
the Phone.
• When LED of the Call Appearance (CA) key on which the call has
arrived, is glowing/blinking. Refer topic ‘Call Appearances’ for
more details.
• Getting the display of the Calling Party’s Number/Name (CLI) on
the LCD.
When user can answer the incoming call?
• When phone is Idle.
• While user is talking to another user.
• When one user is held during a speech to the called party.
• When more than one user is held.
Important Points:
• If the user is in speech with a remote user and if another call
arrives which can be treated as waiting call, Auto Answer will not
be effective. The user will get Call Waiting indication.
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• Auto Answer is effective only if there is no held call and the user is
idle.
• If the user is trying to configure some feature. That means, the
phone is under ‘Configuration’, then the Incoming call cannot be
answered. But if still the user wants to answer the call, he will have
to stop configuring the Phone, exit the SE Mode and should go
OFF-Hook.
• While using headset, if user unplug the headset, call is not disconnected. User can again plug the headset and start conversation.
• Since, the Phone has two DSP channels and remote held call uses
the DSP channel, maximum of two calls can be received if the
same calls are held by remote user after answering the call. Refer
chapter ‘Call Hold’ for ‘Remote Held’.
• However, if there is an incoming/ waiting call when user has two
calls held by remote users, user will still get incoming/waiting call
notification on relevant CA. But, it will not be possible to retrieve
the waiting call since no resources are available to entertain the
waiting call. But to receive the waiting call, user will have to release
one of the calls held by remote user.
• It is possible to answer the waiting call when user has Active
Conference. Maximum two calls can be answered when the user
has active conference.
• When user has active conference which will occupy two CAs and
there is a waiting/ incoming call, user can answer the waiting call
and the two calls involved in conference will be put on ‘hold’. User
will come in speech with the waiting caller.
Relevant Topics:
1. Switching Between Handset, Headset and Speaker 142
2. Making Second Call 99
3. Call Appearances 49
4. Getting Started 27
5. Call Hold 56
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Rejecting a Call
Introduction
• Some times when user is busy with an important meeting and he
gets an Incoming call on his SETU VP236 then he would like to
reject that call.
• Matrix SETU VP236 supports this feature by providing a
programmable key on the Phone, to reject the call.
• It can be programmed from Web page ‘Keys Programming’. Call
can be rejected by menu option, also.
• Assign the key with No LED.
How to Reject a call?
• First answer the required IC/Waiting call.
• Press the ‘Reject’ key to reject all other IC call/Waiting call.
• You can also select ‘Rejecting the Call’ from the ‘Menu’ options to
reject the call.
How it works?
• If there is one IC call then on pressing the ‘Reject’ key, the call can
be rejected.
• If there are more than one IC calls then on pressing the ‘Reject’ key
all the calls will get rejected.
• Hence you have to first answer the waiting call on any CA and then
reject the other calls. For example, If user is in speech on CA1 and
has waiting calls on remaining CA2 to CA4 and user wants to
answer the call on CA2 and reject calls on CA3 to CA4, then he
should first answer the waiting call on CA2 and then use ‘Reject’
feature to reject calls on CA3 and CA4.
• When the phone is ON-Hook and has IC call, remote held call and
local held call, then priority will be given to Remote Held Call, to
answer the call first when user goes OFF-Hook as shown below:
• Priority 1 : Remote Held Call
• Priority 2: IC Call
• Priority 3: Held Call
• A call on CA which is busy can also be rejected.
• After the ‘Reject’ key is pressed or desired option is selected from
the menu, the call is rejected and phone will display normal
condition.
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How to program?
• Click on “Keys Programming” on the Web page assign a Key
Number to ‘reject’ feature. Use this key for rejecting the call.
• Navigate the Menu key and select “Rejecting a Call’ and then
press “Enter” to reject a call. Refer the topic ‘Menu’ to know how to
reject a call from ‘Menu’.
Response during rejecting a call:
The IP Phone will generate and send to remote party, following
responses during rejecting a call:
• For Manual Rejection : Request Terminated.
• For Black Listed caller’s list : Forbidden.
• For Do Not Disturb : Temporarily not available.
• For Anonymous Call Rejection : Forbidden.
• For Call Reject Rejection due to any hardwire limitation : Busy here.
Important Points:
• User must assign a key which does not have LED.
• Do not assign key numbers 06 to 09 for this feature, which have
dual color LED.
• Function of rejecting a call is same while using key or ‘Menu’.
Related Points:
1. Call Appearances 49
2. Menu 100
3. Key Assignment 87
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Rings
Introduction
• When Incoming call comes for the SETU VP236, the user will get
fixed ring cadences as per the country.
• Matrix SETU VP236 supports specific ring cadences for different
countries. Specific ringer tones can also be programmed in the
system.
• Select the Name of country and Ringer Tone Type from the Web
page.
How it works?
• The ring is played over the Phone, as per the ring cadence
specified by the country selected. Refer the table at the end of the
chapter, where ON1 and OFF1 is ON/OFF time for the first
cadence and ON2, OFF2 is the same for the second cadence.
• The tone of the ring will be as programmed.
How to program?
• Click on Web Page “System Parameters”. Refer chapter ‘System
Parameters’ for more details.
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Code Country
01
02
03
04
05
06
07
08
09
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
Australia
Brazil
China
Egypt
France
Germany
Greece
India
Israel
Italy
Japan
Korea
Malaysia
New Zealand
Poland
Portugal
Russia
Singapore
South Africa
Spain
Thailand
UAE
UK
USA/Canada
CADENCE (In Seconds)
ON1 OFF1 ON2 OFF2
0.4
0.2
0.4
2
1
4
1
3
2
4
1.5
3.5
3.5
5.5
0.79
1.1
1
4
0.4
0.2
0.4
2
2
3
1
4
1
2
1
3
2
3
2
3
2
3
1
5
1
3
0.4
0.2
0.4
2
0.4
0.2
0.4
2
1.5
3
2
3
2
3
0.4
0.2
0.4
2
2
4
Remarks:
• Waveform is Sine Wave.
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Search Contact
Introduction
• While using SETU VP236, the user can make a call, after
searching the contact. He can dial out the number by pressing a
key. This advanced feature of the Phone is called “Search Contact’.
• Assign the key with No LED.
• Press ‘Search Contact’ key to display the contacts stored in
the Phone Book as per their names in alphabetical order.
How it works?
• When the user presses any of the dial pad key, the names
matching with the characters of that key is displayed.
• For example, after pressing the ‘Search Contact’ key, the user
presses ‘6’, the contact starting with ‘M’ are displayed. However,
now if the user presses ‘6’ one more time the user will get contact
starting with ‘M’ only and not with N. Refer Table of characters in
chapter ‘Know Your SETU VP236’.
• If the user presses ‘5’ and the contact starting with ‘J’ is not
available in the phone book, the user will get the contacts starting
with ‘K’ and so on.
• When user presses enter key on any of the contacts, the number
stored against the name of the contact is displayed.
• When the user presses ‘Up/Down’ key, other contacts are
displayed.
• Finally user can make a call by going OFF-Hook or pressing the
speaker key/headset key.
Configuring:
• Click on Web page “Keys Programming” to assign a key for this
feature.
• You can also select ‘Search Contact’ by navigating options in
‘Menu’ on Phone.
Related Topics:
1. Phone Book 117
2. Know Your SETU VP236 17
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Selective SIP Line
Introduction
• You can select a specific line and then make an OG Call. This is
called “Selective SIP Line” feature.
• Matrix SETU VP236 supports this feature by providing
programmable keys on the Phone for both the SIP Lines: SIP1 and
SIP2.
• The speech and display on the LCD are just like a normal OG call
which was initiated without selecting SIP1/SIP2.
• Assign a key with single color LED for this feature.
• To use this feature:
• Press the SIP1 or SIP2 key and get the dial tone. Then dial digits
and press “Enter” for OG Call.
• You can also, select the SIP1 or SIP2 option from the Menu, by
navigating the parameters with up/down key and get the dial tone
to dial a number.
How it works?
• When the user presses SIP1/SIP2 key or make a selection using
the options in the menu:
• User gets dial tone.
• Display on LCD for respective CA-row will be as: Dial….
• The LED of the SIP1/SIP2 key glows.
• Display on LCD in the call status characters of the respective CArow will be as: SIP1/SIP2.
How to program?
• Refer chapter Key Assignment, to assign key for SIP1 or SIP2.
• Select ‘Access SIP1 Trunk’ or ‘Access SIP2 Trunk’ by navigating
‘Menu’ on phone.
Related Topics:
1. Key Assignment 87
2. Menu 100
3. OG Call Route Selection 110
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SIP Parameters
Introduction
• Matrix SETU VP236 supports a feature “SIP Parameters’, by which
some parameters of SIP trunks like registrars, Port addresses can
be programmed.
• It can be programmed by Web Page. But some parameters like SIP
ID, Registrar Server Address, etc. are also programmable from
Phone so that the user can start using the Phone without the need
of a computer to configure it.
• Refer Glossary and VoIP Basics for explanation of Voice
Compression (codecs), SIP, RTP, STUN, OUTBOUND Proxy, etc.
How it works?
Matrix SETU VP236 supports the SIP parameters:
• Configurable by Web
• Configurable by Phone
Parameters configurable by Web using SIP1 and SIP2:
• SIP Account-Enable/Disable
• Display Name
• SIP ID
• Registrar Server Address
• Registrar Server Port
• Re-Registration Timer
• Registration Retry Timer
• Authentication User ID
• Authentication Password
• Vocoder Preference
• DTMF Option
• Outbound Proxy-Enable/Disable
• Outbound Proxy Server Address
• Outbound Proxy Server Port
• Use STUN?-Yes/No
• SIP parameters programmable from phone from ‘SIP1
Parameters’ or ‘SIP2 Parameters’ after entering ‘Config Phone
Menu’.
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•
•
•
•
•
•
SIP Status
SIP ID
Registrar Server Address
Registrar Server Port
Authentication User ID
Authentication User Password
How to use?
• For programming by the Phone, Navigate Menu → Config Phone
→ SIP1 Parameters.
• Refer Web Jeeves. Click on Web page “SIP Parameters”.
Registrar Settings:
Program following parameters for both SIP1 and SIP2.
• Status:
It shows which SIP line is enabled/disabled:
• Select ‘enabled’ if you want to make OG calls/receive IC calls
from this SIP line. If it is enabled, corresponding characters ‘SIP1’
or ‘SIP2’ will be flashing on the LCD display, with a cadence of
ON/OFF.
• Select ‘disabled’ if you do not want to allow SIP line to make OG
calls even if it gets selected by the OG call routing logic. Also IC
calls will not be received.
• By default, Status is Disabled.
• Display Name:
Enter here the name which you want the called party to receive on
his Phone.
Maximum 24 characters can be entered. ASCII characters can be
used. By default, Display Name is Blank.
• SIP ID:
SIP ID is a full SIP URI. This can be a number or text. Maximum of
24 characters can be entered with all ASCII characters. A SIP URI
provided by the ITSP can be [email protected] or [email protected] .
When the user part is a text, the SIP ID of a user is distinguished
from his mail ID or any other ID by a SIP Tag. i.e. SIP:
[email protected]. By default, SIP ID is Blank.
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• Registrar Server Address:
SIP Registrar Servers’ Address is the address of the SIP Registrar.
It can be IP address also. This may same as the SIP Server
address since the SIP Server might also be acting as Register
Server.
40 characters maximum with the extended ASCII characters.
By default, Registrar Server Address is Blank.
• Registrar Server Port:
SIP Registrar Servers’ Port is the Registrar Server’s listening port
for SIP. User may change this field if the ITSP provides a SIP port
number other then the default.
Valid range = 1024-65535. This may be same as SIP Server’s Port
address.
By default, Registrar Server Port is 5060.
• Re-Registration Timer:
• Normally, the Register server deletes an entry of its client from its
database on expiry of a fixed timer. This timer is set by the
Register Server. Hence, in order to be registered always, the IP
Phone should send a Registration request before this timer
expires.
• This timer signifies the time after which the SETU VP236 should
send Registration request again to be registered. To be on safe
side as a rule, the SE should program this time = half of the
Registration time set by the Register Server. Suppose, the
Register Server deletes the entry after 30 minutes of registering a
client. In such case, the SE should program this time=15minutes.
Re-Registration Timer range is from 00001 to 65535.
By default, Re-Registration Timer is 3600 seconds.
• Registration Retry Timer:
On expiry of Re-Registration Timer, send the registration request. If
the registration attempt fails, send the registration request on
expiry of this timer. Phone will send the registration request till IP
Phone gets registered and once it gets registered again ReRegistration Timer will be started.
Registration Retry Timer range is from 00001 to 65535.
By default, Registration Retry Timer is 10 seconds.
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Authentication
• Authentication User ID:
This is the user name for registering the SIP account with the SIP
register server.
Enter maximum of 16 characters with all ASCII characters.
By default, Authentication User ID is Blank.
• Authentication Password:
This is the password associated with the user name above.
Maximum of 16 characters can be entered with all ASCII
characters.
By default, Authentication Password is Blank.
• Vocoder Preference:
• To compress bandwidth of data in the RTP Packets, various
codecs are used.
• As per optimization criteria of bandwidth and speech quality, the
coders are supported by the phone like G.722, G.723, etc.
• While call is established, the vocoder is selected as per
autonegotiation with remote SETU VP236. This parameter
signifies the Vocoders in preferential order which can be
requested to the remote user while making OG call.
• In case of IC calls, the preference requested by the remote end
will be considered. If none of the Vocoders requested by the
remote end is supported by user, the call is dropped and logged
as rejected call in Call Logs.
Select the vocoder from the options shown below:
For SIP1 and SIP2 Default Vocoder Preferences
1st Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
2nd Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
3rd Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
4th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
5th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
6th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
7th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
8th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
9th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
10th Pref.
G.729
G.723-5.3
G.723-6.3
G.722
G.726-16
G.726-24
G.726-32
G.726-40
PCMA
PCMU
The user may select all the ten preferences to be same.
Default Vocoder is shown as bold.
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For Example:
If the user selects G.729 in all the preferences then if the remote user
supports this Vocoder, then only it will accept the request and the call
will be through. But if the remote user does not support the request,
the call shall be rejected.
• DTMF Option:
This parameter will decide how the DTMF digits will be sent over
SIP.
This option is used while making OG calls. In case of IC calls, the
option requested by the remote end will be considered.
Three options are provided.
• RTP (RFC 2833)
• SIP (INFO)
• In-band
By default, DTMF option is SIP (INFO).
Outbound Proxy:
Status:
Select from Yes/No if you want to enable Outbound Proxy and if
your ITSP has this server to handle voice calls.
By default, Outbound Proxy is No.
• Outbound Proxy Server Address:
This specifies the Outbound Proxy Servers’ Address. It can be IP
address also. This may be same as the SIP Server address since
the SIP server might also be acting as Enter here the address of
Outbound Proxy of 40 characters maximum It can be with
Extended ASCII characters.
By default, Outbound Proxy Server Address is Blank.
• Outbound Proxy Server Port:
This specifies the Outbound Proxy Server’s listening port for SIP.
User may change this field if the ITSP provides an Outbound Proxy
Server port number other than the default. This may be same as
SIP Server’s Port address.
Enter from the Range = 1024-65535.
By default, Outbound Proxy Server Port is 5060.
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STUN
Use STUN:
This parameter specifies whether STUN will be used for this SIP
trunk.
Select “Yes” if STUN is to be used to get the IP address.
Select “No” if STUN is not to be used.
By default, STUN is No.
Important Point:
• Refer “Network Port” for other parameters like, IP address and the
port number for the STUN server.
Relevant Topics:
1. Configuring SETU VP236 16
2. Web Jeeves 168
3. Know Your SETU VP236 17
4. Network Port 102
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Software Upgrade
Introduction:
• Matrix SETU VP236 supports to upgrade the IP phone for desired
config files, drivers, firmware, call logs, html files using FTP server.
• File Transfer Protocol (FTP) is used for exchanging files over LAN/
WAN.
• It is the most frequently used protocol to download file from a FTP
server or to upload file to the FTP server.
• FTP server is also required to make desired configuration files
which can be downloaded using Auto Configuration feature which is
used by ITSPs.
• For Example, if user purchases the IP phone from ITSP, ITSP will
be administrator of IP phone and customer will be user of IP phone.
It is assumed that ITSP will not reveal config password to their
customer to restrict them to use some feature/settings. It is also
assumed that ITSP will retain the customer wise record of config
password of IP phone units for future reference. In case, ITSP
engineer needs to configure/repair IP phone at customer site and
forgot config password, then he can make the password
configuration file (using feature ‘Software Upgrade’) using another
IP phone and copy/paste that file into the user folder stored in
TFTP server. Now, after restarting the IP phone, IP phone will
download the configuration files from TFTP server using ‘Auto
Configuration’ and update the config password to the desired one
as changed by ITSP.
How to use it?
• SE/Admin can view and modify the relevant files of Firmware,
Drivers, Web Pages, Configuration and Call Logs.
• Click on ‘Software Upgrade’.
• You will get the FTP window which displays the following folders:
• Appl
•
Clicking ‘Appl’ folder will display following files:
•
Mxipphone-Main application firmware
•
Mxhttpd-Web Server
•
Boardcnfg-Configuration of Ethernet interface
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• Driver
•
Clicking ‘Driver’ folder will display the following files:
•
Hapi.o-DSP driver
•
Mxlcd-LCD driver
•
Mxlcd.o-LCD driver
• Webpage
•
Clicking ‘Webpage’ folder will display all the html files of Web
Jeeves.
• Config
•
Clicking ‘Config’ folder will display all the configuration files
which IP phone is currently using.
• Logs
•
Clicking ‘Logs’ folder will display all the call logs files.
•
These files are required in typical conditions when user wants
to store the few call logs for future reference.
• Clicking any folder will display the relevant files which IP phone is
currently using.
• SE/Admin can cut, copy and paste desired files here.
• Finally new configuration/ driver/ firmware will take effect only
during next power ON.
Relevant Topic:
1. Auto Configuration 44
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Speed Dial
Introduction
• Using this feature it is possible for the user to dial a number simply
by pressing a key instead of pressing long digits of the number
string of the Phone Book.
• Assign a key with No LED from web page.
How to use it?
• Press the key which is assigned Index Number of ‘Phone Book’.
The number string programmed in the phone book will be dialed out
by the phone.
How it works?
• SETU VP236 has a database of contacts viz. Phone book. This
database is used to implement this feature.
• While configuring the keys, the user may select Phone Book as a
feature for the key and the respective Index in the address book.
• Since SETU VP236 supports a maximum of 18 programmable
keys, maximum of 18 phone numbers/addresses among the
address book entries can be programmed as speed dial numbers.
• When the user presses the speed dial key, the number is displayed
on the LCD and the call is made.
• If the user programs an index of the address book which does not
have a number, then he will get display ‘No Number Programmed’
and will get error tone if that number is dialed.
How to program?
• Ensure that ‘Phone Book’ is programmed as required.
• Refer the topic ‘Web Jeeves’ for more details. Click on ‘Key
Programming’.
• Select ‘Key Number’ which has no LED for which ‘Speed Dial’
feature is required.
• Feature: Select ‘Phone Book’ in this field.
• Phone Book Index Number: Select Index Number of the number
to be dialed, as per Phone Book.
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Relevant Topics:
1. Key Assignment 87
2. Phone Book 117
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Status
Introduction
• Using ‘Status Page’ you can have quick glance for some Network
Parameters and SIP Parameters. You can change it from
respective page, if required.
• It also provides ‘status’ and ‘statistics’ parameters of Auto
Configuration.
• You can also get the status on the display of phone, by accessing
‘Network Status’ from the ‘Menu’.
• Click on ‘Status’ page. Following basic parameters are displayed:
Parameters
Display
Description
This displays the name of
System Name
Matrix SETU VP236
system
This displays the current
Version/
VxRy
version 'x' and Revision 'y'
Revision
of software
Netw ork Parameters
IP Address
192.168.001.161
This displays the static
address or address
located by DHCP or
PPPoE
Subnet Mask
255.255.255.0
This display the
programmed subnet mask
Gateway
Address
DNS Address
DNS Domain
Name
NTP Address
138
This displays the
programmed gateway
address
This displays the
programmed Domain
Name Server Address
This displays the 'Name'
of the Domain Name
Server.
This displays the
programmed SNTP server
Address used by phone
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MAC Address
Unique Address for
the IP Phone
This displays the MAC
Address for the Network
Port programmed at
factory
NAT Type
Blank display or
Unknown or full
cone or Restricted
or Port Restricted or
Symmetric or
Symmetric Behind
Firewall or Blocked
NAT is configured for
option as shown. (1) If
STUN is not enabled, it is
Blank (No Display) (2) if
none of the type of NAT is
detected by VP236, then
'Unknown' is displayed.
Pub lic IP
N.N.N.N.
N = 001 to 255
W hen STUN is enabled
'Public IP' is displayed
here.
SIP Account 1:
Registration
Status
Not Registered/
Registered
Registration
Time
Registration
Retry Count
0,1,2……
SETU VP236 V1 System Manual
W hen 'Register' button is
pressed the IP Phone
sends registration request
to the server and the
'Status' is displayed, here
as 'Registered' or 'Not
Registered'.
This time is provided by
server during registration
and displayed here
This shows how many
time after failure, IP Phone
sent registration request.
As retry count increase
this number displayed as
increasing 0,1,2,…. After
the page submit this will
get reset to '0'.
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This displayed the reason
of last time failure. For
example, Message sent
failure. Response timeout.
Failed response 486, etc.
Registration Last
Failed Reason
SIP Account 2:
Registration
Status
Not Registered/
Registered
Registration
Time
Registration
Retry Count
Registration Last
Failed Reason
140
0,1,2……
When 'Register' button is
pressed the IP Phone
sends registration request
to the server and the
'Status' is displayed, here
as 'Registered' or 'Not
Registered'.
This time is provided by
server during registration
and displayed here
This shows how many time
after failure, IP Phone sent
registration request. As
retry count increase this
number displayed as
increasing 0,1,2,…. After
the page submit this will
get reset to '0'.
This displayed the reason
of last time failure. For
example, Message sent
failure. Response timeout.
Failed response 486, etc.
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Auto Configuration
Status
Blank
Statistics
Blank
Message displays after
Auto Configuration is
applied during Power ON:
for e.g. 'No tftp Server
Specified', 'Auto
Configuration Fail', 'Auto
Configuration Success'.
Message displays after
Auto Configuration is
applied: for e.g. 'Files
Downloded/Configured
=X/Y'.
Relevant Topics:
1. Network Port 102
2. SIP Parameters 128
3. Auto Configuration 44
4. Menu 100
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Switching between the Handset, Headset and
Speaker
Introduction
• Matrix SETU VP236 supports talking and listening by Handset,
Headset and Speaker/MIC. This feature allows you to switch
between handset, headset and speaker during the call.
• Speaker mode or Headset mode can be used by pressing
respective key.
• ‘Spk’ key is with LED indication.
• The number for Headset feature is configurable by Web page “Key
Assignment’.
How it works?
• When using handset: (To switch to Headset and Speaker)
• Press “Headset” key to switch to headset Mode Speech audio is
connected to the Headset port. User can go ON-Hook irrespective
of headset is connected or not.
• Press the “Speaker” key to switch to Speaker Mode. The Speaker
is turned ON. Replace the handset on the cradle.
• When using headset: (To switch to Handset and Speaker)
• Lift the handset to use handset. While using headset, if user goes
OFF-Hook, the voice will be available on the handset port.
• Press the “Speaker” key to use speaker. The headset port is
disabled and the speaker is turned ON.
• When using speaker: (To switch to Handset and Headset)
• Lift the handset to use handset. While using speaker if user goes
OFF-Hook, the speaker is turned OFF and the voice will be
available on handset port.
• Press “Headset” key to use headset. While user presses
“Headset” key, the voice will be available on the headset port and
the speaker is turned OFF irrespective of headset is connected or
not.
How to Configure?
• Click on Web page “Keys Programming’.
• Assign a key to ‘Headset’ function.
• Refer chapter ‘Key Assignment’ for more details.
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Important Points:
• In headset mode, if user unplugs the headset, call will not be
disconnected. User can again plug the headset and start the
conversation on headset.
• Using Headset:
To configure the phone to operate in ‘Headset Mode’, the user can
select the feature ‘Headset ON-Hook/OFF-Hook from the menu.
• Using the Speakerphone:
To configure the phone to operate in ‘Speakerphone Mode’ the
user can press ‘spk’ key.
Related Points:
1. Making a Call 94
2. Receiving a Call 119
3. Key Assignment 87
4. Menu 100
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System Parameters
Introduction
This chapter describes important system parameters used by the
SETU VP236 for its satisfactory operation. Refer corresponding
chapter for more details.
How to use Web-page?
Click on Web Page “System Parameters” and program the parameters
as shown below:
Parameter
System Name
CPTG Type
Selected
country
End of dialing
Enable End of
dialing digit
Value/Default
Max. 24 char.
Default Matrix
SETU VP236
Description
Enter here the name of phone
you are using.
List of countries
Default = India
Select the ‘country name’ from
the list of countries for countryspecific CPTG.
Selected/Blank
Default=Selected
(Enable)
Enable Inter digit Selected/Blank
wait timer for
Default=Selected
End of Dialing
(Enable)
Do not select if you want to
disable end of dialing feature.
Do not select if you want to
disable ‘Inter digit Wait Timer”
feature.
Fixed Number of Selected/Blank Select if you want that system
digits for End of Default=Selected should dial out the number after
entering programmed fixed,
(Disable)
dialing
number of digits.
---do----1-40
This shows that system
Default = ‘40’
considers end of digit after
receiving ‘40’ digits of number
string.
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Parameter
Keypad Lock
Auto keypad
Lock
LCD Settings
Backlight
Value/Default
Set / cancel
Select ‘Set’ if you want that your
Default = Cancel Phone keypad can not be
operated after a time of ‘Keypad
lock timer’ when Phone is in Idle
state. You need not press key
for ‘keypad lock’. Else select
‘Cancel’.
Select the option as per your
Always ON,
Always OFF, For requirement for LCD setting.
10 sec. (Default),
20 sec., 30 sec.,
60 sec., 120
sec., 240 sec.
Brightness
1- 4, default = 2
Contrast
1- 4, default = 2
Number of Call Appearances
No. of CA’s
1- 4, default = 4
Ring Type
Ringer Tone
Type
Select country
Description
0-9, Default =
Ttype 6.
List of countries
Default = India
SETU VP236 V1 System Manual
Select level from 1 to 4. With ‘1’
it is lowest brightness for
suitable viewing
Select level from 1 to 4. With ‘1’
it is lowest contrast for suitable
viewing
Select number as per your
requirement of indication for
waiting calls on the LCD. For
indication of more IC calls,
reject one of previous calls.
Select the tone type of ring for
IC call
Select the country name from
list of country.
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Parameter
Value/Default
Volume Setting
Handset Tx
0-7, Default = 4
Description
Select volume of transmit
speech as per your desired
level.
Select volume of received
speech as per your desired level
Handset Rx
0-7, Default = 4
Headset Tx
0-7, Default = 4
Select volume of transmit
speech as per your desired level
Headset Rx
0-7, Default = 4
Select volume of received
speech as per your desired level
Microphone Tx
0-7, Default = 4
Select volume of transmit
speech as per your desired level
Speaker Rx
0-7, Default = 4
Select volume of receive speech
as per your desired level
Ringer
0-7, Default = 4
Select volume of incoming ring
as per your desired level
Timers
ACB notification
Timer
Auto Answer
Timer
01-99, Default =
30 Min.
01-99, Default =
02 sec.
Call Forward
Timer
Call Hold Alert
Timer
First Digit Wait
Timer
146
Refer chapter ‘Timers’ for
definition
Enter time for which Phone will
wait before auto answering the
IC Call
00-99, Default = Refer chapter ‘Timers’ for
45 sec.
definition. Refer chapter ‘Call
Forward’
000-999, Default Refer chapter ‘Timers’ for
= 180 sec.
definition
01-99, Default = Refer chapter ‘Timers’ for
15 sec.
definition
SETU VP236 V1 System Manual
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Parameter
Hotline Timer
Value/Default
0-9, Default = 0
Keypad Lock
Timer
01-99, Default =
15 sec.
Inter Digit Wait
Timer
Ring Timer
01-99, Default =
06 sec.
01-99, Default =
45 sec.
Transfer
Notification
Timer
01-99, Default =
60 sec.
Description
Enter here time delay after
which you want your phone to
dial the hotline address when
OFF-Hook.
Enter time after which keypad
will be Automatically locked.
Key can be unlocked by
pressing assigned key.
Refer chapter ‘Timers’ for
definition
Enter time up to which 'Ring' will
be played after IC call comes for
your Phone.
Refer chapter ‘Timers’ for
definition
Relevant Topics:
1. Timers 150
2. Auto Answer 40
3. Hotline 85
4. Call Hold 56
5. LCD Settings 92
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Time Settings
Introduction
• For the Matrix SETU VP236, the LCD display consists of Date and
Time. Also some features like Call Logs use the actual time of call
etc. But the Phone does not have Hardware design for the RTC
(Real Time Clock). Hence the SETU VP236 needs to be up dated
automatically for correct time at fixed time intervals.
• The SETU VP236 provides feature for time setting by using Simple
Network Time Protocol (SNTP) to get time from the Time Server.
• Refer NTP in ‘Glossary’.
How it works?
• User has to program the time server in the Web page.
• Phone will get the time from the NTP server every 10 seconds
(approx).
• The time server provides GMT (Greenwich Mean Time).
• The SETU VP236 is located at the country which is situated in the
specific time zone.
• Based on the time zone settings, Matrix SETU VP236 shows
correct time for that country. For example, if time shown by
ntp1.cs.wisc.edu server for India (Culcutta, Chennai, Mumbai, New
Delhi) is 10:44:50AM, then current time shown for Kabul is
09:44:50AM. Because for India GMT is ‘GMT +5.30’ but for Kabul it
is ‘GMT +4.30’ (1 hour behind India). The date is displayed such as,
17May2006.
• By default following three Time Servers are programmed:
• Ntp1.cs.wisc.edu
• Time.windows.com
• Time.nist.gov
• You can select any one option among three to obtain the date and
time or may enter NTP Server address of your own choice.
• The countries for “Time Zone” supported are same as supported by
Windows. By default selection is Calcutta, Mumbai, Chennai
and New Delhi.
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How to use Web Page?
Click on Web page “Time Setting” and program Time Server and
Time Zone. Refer Web Jeeves.
Parameter Value/Default
Description
Phone will display here the
Current
DD-Month namecurrent date available from
Date
YYYY.
Server
Default = 01
January 2007
Current
HH:MM:SS. Default Phone will display here the
Time
= 10:00:00
current time available from
Server
Select suitable Time server or
NTP
Ntp1.cs.wisc.edu
Address
Time.windows.com Enter the desired NTP server
address for ‘time’ display on
Time.nist.gov
the Phone
default =
Ntp1.cs.wisc.edu
Time Zone List of places as per Select the place name as per
the country where Phone is
MS Windows.
Default = Calcutta. working.
Relevant Topic:
1. Daylight Saving Time (DST) 76
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Timers
Introduction
• SETU VP236 supports various kinds of timers. For example, if user
waits for some period after entering digits then instead of expecting
required number of digits, system will automatically dial out the
number string after a fixed time interval. This time is called “Inter
Digit Wait Timer”.
• Matrix SETU VP236 is designed with following types of timers:
• ACB Notification Timer
• Transfer Notification Timer
• Auto Answer Timer
• Ring Timer
• First Digit Wait Timer
• Inter Digit Wait Timer
• Call Hold Alert Timer
• Call Forward Timer
• Hotline Timer
• Key Pad Lock Timer
Explanation:
• Notification Timers: These timers are related to response from
the phone while setting a feature.
• Transfer Notification Timer: This timer signifies the time for
which the SETU VP236 should wait for the notification of transfer
function from the transferee. This timer is programmable.
• ACB Notification Timer: This timer signifies the time for which
the SETU VP236 should wait for notification for ACB Subscribe
ACB ‘ready for call completion’ notification. This timer is
programmable.
• Ring Timer: This timer signifies the time for which the SETU
VP236 will give ring indication, when incoming call comes. When
ring will be stopped the Phone will go to idle state.
• First Digit Wait Timer: This timer signifies the time for which the
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SETU VP236 should wait for first digit. On expiry of this timer,
phone will go to idle state or standby state. This timer is
programmable.
• Inter Digit Wait Timer: This timer signifies the time for which the
SETU VP236 should wait between two digits. On expiry of this
timer, the number is considered to be complete and is taken up for
further processing. This timer is programmable.
• Call Hold Alert Timer: This timer signifies the time after which the
SETU VP236 will remind the user about held call. This timer is
programmable.
• Call Forward Timer: This timer signifies the time after which SETU
VP236 will forward the call to programmed destination number.
This timer is programmable.
For other timers, refer chapter ‘System Parameters’.
Programming:
All the timers are programmable from Web only. Click on the Web
page ‘System Parameters’.
Relevant Topic:
1. System Parameters 144
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User Settings
Introduction
• This chapter explains how you can set the features which are
frequently used, like Auto Answer, CLIR, etc. using Web page. It is
also possible to set some of the features using ‘Menu’ key available
on the SETU VP236.
• Refer corresponding chapter for more details.
• Refer ‘Web Jeeves’. Click on ‘User Setting’.
Which features can be configured?
Feature
Description
Anonymous Call Select ‘Set’ or ‘Cancel ‘. Default = Cancel
Rejection
Auto Answer
Select ‘Set’ or ‘Cancel ‘. Default = Cancel
Black Listed
Enter Number from which you do not want to
Caller List
receive a call. Max. 10 numbers can be entered.
with 40 characters max. Default = blank
Call Forward
Refer chapter Call Forward to program the
features: CF-Busy and CF-No Reply.
CLIR
Select ‘Set’ or ‘Cancel ‘. Default = Cancel
Do Not Disturb Select the required option : DND All, DND List,
Cancel. Default = Cancel.
DND List
Enter a number which you want to allow as
Incoming call on your Phone even if you have set
DND feature. Upto 40 characters can be entered.
May be with ASCII. Max.10 numbers can be
entered. Default = Blank
Hotline
Refer chapter “Hotline’
Hotline Address Refer chapter “Hotline’
Voice Mail Account1
VM Subscription Enter Max. 40 characters for VM subscription
Address
address-1 available from ITSP. It can use all
ASCII characters. Default = Blank.
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VM Retrieval
Address
Enter Max. 40 characters for VM retrieval
address-1 available from ITSP. It can use all
ASCII characters. Default = Blank
Voice Mail Account2
VM Subscription Enter Max. 40 characters for VM subscription
Address
address-2 available from ITSP. It can use all
ASCII characters. Default = Blank.
VM Retrieval
Address
Enter Max. 40 characters for VM retrieval
address-2 available from ITSP. It can use all
ASCII characters. Default = Blank
Relevant Topics:
1. Call Forward 52
2. Hotline 85
3. Voice Mail 157
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VLAN Tagging
Introduction
• LAN is defined as a single broadcast domain with Standard (IEEE
802.1Q). VLAN is Virtual LAN.
• If a user broadcasts information on his/ her LAN, the broadcast will
be received by every other user on the LAN. VLANs allow to
logically segmenting a LAN into different broadcast domains.
• In order to extend VLANs across different switches, a link called
‘trunk’ must interconnect the switches.
• Now, when you want traffic from multiple VLANs to be able to cross
a link that interconnects two switches, there is a need to configure
a VLAN tagging method on the ports that supply the link. A frame is
tagged when it leaves the switch with information about the VLAN
that the frame belongs to.
How it works?
• IP phone has three port inbuilt switch. These ports are CPU port,
Network port and PC port. Network port of IP phone will be
connected to the external switch in which multiple VLANs will be
configured.
• If you enable the VLAN, the IP phone will tag all the packets leaving
the Network and PC port and also receives only packets on PC port
and Network port with VLAN tag.
• Now, if you enable the VLAN by mistake and PC/ switch to which IP
phone is connected does not support VLAN tag, you will not be
able to access the Jeeves for further configuration. This is because
the Phone will expect only tagged packets and ignore all the
packets which are untagged.
• Hence the Phone supports to enable/disable the VLAN, from the
Jeeves as well as ‘Configuring the Phone-Menu’, so that in above
case if Jeeves can not be accessed, VLAN can be disabled from
the ‘Configuring the Phone-Menu’ and then Jeeves can be used for
further configuration.
For Example:
• The example will be useful for programming of the feature.
• Suppose 4 VLANs are configured in external switch, VLAN1
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(Marketing), VLAN2 (Accounting), VLAN3 (R&D), and VLAN4 (HR).
To limit the broadcasting only to marketing department, VLAN Id
configured in IP phone will be ‘1’. Now, when the frames will leave
from the network port of IP phone, IP phone will add the VLAN tag
in frames with VLAN Id1. When these frames will reach the port of
external switch/ router to which IP phone is connected, it will read
the VLAN header and identify that frame belong to VLAN1
(marketing) and will forward the frame to VLAN1.
• In above example, if you want to shift the user who is using IP
phone to the other department say HR, just change the VLAN Id to
4 and IP phone will tag the frames leaving network port with VLAN
Id 4.
• Thus following parameters are required to be programmed for this
feature:
• Option for using (adding) VLAN Tag-Yes or No.
• VLAN ID.
How to program?
• Refer Web Jeeves.
• Click on ‘Network Parameters’.
• The option to use VLAN tag or not (Yes/No) and VLAN ID, can be
programmed from web page.
• Refer chapter ‘Network Port’ for programming details.
• ‘VLAN Setting’ can be enabled/disabled by navigating:
• Menu > Config Phone Menu > Network Parameters > VLAN
Setting. ‘VLAN Setting’, provides two parameters:
• Status
• VLAN ID
Select ‘Status’ to access enable/disable.
Select ‘Enable’ to use VLAN feature.
• After selecting ‘VLAN ID’, program it from range 0 to 4095.
Default=0.
Important Point:
• IP phone will not tag the packets coming from PC port and leaving
the network port of the 3-port switch built-in to the Phone.
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Relevant Topics:
1. Network Port 102
2. Configuring the Phone-Menu 75
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Voice Mail
Introduction
• SETU VP236 supports to notify you about a new message in your
mailbox. (This is possible only if it receives a notification from the
Voice Mail Server from ITSP). This feature is called “Voice Mail”.
(VM) Matrix supports two VM servers.
• You can also access the mailbox by selecting a user option (Menu).
• You can assign separate key with single color LED to access VM
Account1 or VM Account2. Two Voice Mail Server Address are
programmable from Web, also.
How to use it?
To access the mailbox:
• Press a separate key assigned to access Voice Mail Account1 or
Voice Mail Account2.
• You can also, select the option ‘Call VM Server1’ or ‘Call VM
Server2’ from the Menu.
How it works?
• This feature is implemented with two requirements viz. Voice
Message Notification and Voice Message Retrieval. Voice
message Notification is by the LED whereas, Voice Message
Retrieval is through any of the two programmed keys.
Voice Mail notification from VM Server
• You can get the voice mail notification message from VM servers in
any of the port states of the Phone.
• On receipt of notification message from either or both VM servers:
• LED of the key will glow Red.
• The message on the LCD will be displayed.
• Default VM key is for ‘VM Account1’.
Voice Mail retrieval from VM Server:
• User can program the separate keys for VM Account1 and VM
Account2 from web jeeves.
• For each voice mail account following two parameters can be
programmed from Web page:
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•
•
•
•
• VM Subscription Address
• VM Retrieval Address
The system will send the VM subscription request to the address
programmed for the ‘VM Subscription Address’ for the
corresponding VM account.
You can retrieve the voice mail/messages:
• From Menu
• Using a key
On pressing any of the separate key for ‘Call VM Server1’ or ‘Call
VM Server 2’:
• If there is no notification from VM servers, the display will be as
‘Call VM Server1’ & ‘Call VM Server2’. You can call the selected
Server.
• If there is notification from one of the VM servers say from VM
Server1, on pressing ‘Call VM Server1’ key, the Phone will display
‘New message in VMS1’ & ‘Call VM Server2’. If you select the
first option, voice mail server1 will be called and the LED of the
key will be turned OFF, the display will be changed to ‘Call VM
Server1’. Also, displayed notification message on LCD will be
removed.
• If there is notification of new mail/message from both the VM
servers, the Phone will display ‘New message in VMS1’ & ‘New
message in VMS2’. If you select one of the option say VM
server2, VM Server2 will be called. Now if user selects ‘New
message in VM Server1’, VM Server1 will be called. LED will
be turned off and display will be changed to ‘Call VM Server1’.
Also, displayed notification message on LCD will be removed.
If a key is not assigned, then you can access voice mail box from
the Menu by selecting the option ‘Call VM Server1’ or ‘Call VM
Server2’.
Configuring:
Following two parameters can be configured to make this feature
work:
• Programming of the Voice Mail Server Address. Refer topic ‘Web
Jeeves’, page “User Setting”. Refer chapter ‘User Settings’ for
programming of parameters.
• For assigning a key. Refer topic ‘Key Assignment’. Assign separate
key with single color LED to ‘VM Account1’ and ‘VM Account2’
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feature.
• To access from Menu:
• Select the option ‘Call VM Server1’ or ‘Call VM Server2’ and
press ‘Enter’ key.
• System will dial out voice mail server address.
Important Points:
• All the features like hold, transfer, conference, etc. applicable to a
normal call are applicable in this case also as this call is also
considered as a normal call (except that it is made to a Voice Mail
Server and not to a human).
• Follow the prompts for the voice mail system to listen to and delete
your voice mails.
Relevant Topic:
1. User Settings 152
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Volume Settings
Introduction
• SETU VP236 supports to increase or decrease the speech audio
level of the call.
• You can set:
• Handset Transmit Volume: So that your voice is heard clearly
by the remote user while using handset.
• Handset Receive Volume: So that the remote user’s voice is
heard clearly by you while using the handset.
• Headset Transmit Volume: So that your voice is heard clearly by
the remote user while using headset.
• Headset Receive Volume: So that the remote user’s voice is
heard clearly by you while using headset.
• Speaker Transmit Volume: So that your voice is heard clearly by
the remote user while using speaker.
• Speaker Receive Volume: So that the remote user’s voice is
heard clearly by you while using speaker.
• Ringer Volume: So that you can hear clearly the ringing cadence
played by the phone.
• Feature can be configured from Web and Menu options.
How to use it?
• To adjust handset transmit volume:
• Go to “Config Phone Menu”, select the option ‘Volume Settings’
• Select option ‘Handset Transmit Volume’
• Enter any value between 0 to 7 to get desired level. (you can
confirm the effect by asking the remote user).
• To adjust handset receive volume:
• Go to “Config Phone Menu”, select the option “Volume Settings”.
• Select option “Handset Receive Volume”
• Enter any value between 0 to 7 to get desired volume level.
• Similarly use the feature for headset and speaker options.
How it works?
• You can adjust volume settings while in speech or when idle. If in
speech, the speech will get effect of increase/decrease. If set in
Idle state of phone, you can feel the level by picking up the
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handset or turning ON the speaker or headset.
• The Phone supports levels of Volume from Level 0 to Level 7.
(Level0 being lowest and Level7 being highest).
• You can feel the setting of Ring Volume only when you receive the
Incoming Call, when your phone is NOT in Idle condition.
• But if your phone is in ‘Idle’ condition, you can feel the setting when
RBT (Ring Back Tone) is played.
• You can set volume during different condition of using the phone as
mentioned below:
• When using Handset:
•
Press ‘Å/Æ’ key on the Phone, to increase/decrease Rx gain
of speech/call progress tones respectively.
• When using Headset:
•
When you are in speech over headset, your Rx level of the
headset can be changed in same way as explained above.
This will be required if remote user has not set his Tx audio
level properly.
• During Ringing:
•
Follow same method while the ringer is ringing on receipt of
Incoming call and listening it.
Configuration:
• To configure the feature, Click on Web Page “System
Parameters”.
• To configure from Phone: Navigate Menu > Config Phone Menu >
Volume Setting.
Relevant Topics:
1. System Parameters 144
2. Menu 100
3. Call Progress Tone Generation 65
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VoIP Basics
VoIP is the sending of voice signal over Internet Protocol. Calling on
VoIP allows user to make phone calls and send faxes over the internet
at a negligible cost compared to the normal circuit switched Telephone
network. The main components of VoIP are:
• SIP
• Network Servers
• RTP
• NAT
• STUN
• Outbound Proxy
• SIP Call Flow
• Important Voice Parameters: Jitter Buffer, Voice Compression and
QoS.
SIP:
• The signaling protocol which is used to establish an IP call is called
SIP Session Initiation Protocol. It is an application layer protocol
that handles the setting up, altering and tearing down of voice and
multimedia sessions over the internet. Refer RFC 3261 for more
information.
• SIP handles telephone calls and can interfere with PSTN network.
• A SIP account uses complete identity called, URI (Uniform
Resource Identifier) or SIP Address, like e-mail address and is
given by: SIP-Number@SIP service Domain.
• SIP number can use letters or numbers like ajay@ITSP
Provider.com or 73121@ITSP Provider.com. A SIP service domain
is the domain name in the SIP URL. For example, if SIP address
is 73121@VoIP Provider.com, then ‘VoIP Provider.com’ is the SIP
service domain.
• In general, SIP follows the Client Server Architecture. To support
this, there are two main entities user agent and network servers.
The peers in a session are called User Agents (UAS). A user
agent can function in one of the following roles:
• User Agent Client (UAC)-A client application that initiates the
SIP request.
• User Agent Server (UAS)-A server application that contacts the
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user when a SIP request is received and that returns a response
subject to user’s input.
Network Servers:
There are three types of network servers in a SIP network: Proxy
Server, Redirect Server and Registrar Server.
Proxy Server: A SIP proxy receives a request, makes a determination
about the next server to send it to, and forwards the request, possibly
after modifying some of the header fields. As such, SIP requests can
traverse many servers on their way from UAC to UAS. Responses to
a request always travel along the same set of servers the request
followed, but in reverse order.
Redirect Server: The redirect server does not forward requests to the
next server. Instead it sends a redirect response back to the client
containing the address of the next server to contact.
SIP Registrar: SIP Registrar is an entity where SIP users can get
themselves registered. Registrar imparts mobility to the SIP users. A
SIP user can register himself with a registrar. If the user changes his
location, he has to register again with the registrar stating his latest
contact information. Whenever a call is to be delivered to that user,
Registrar can provide the information about the location where the
user was active recently.
RTP:
After a VoIP call is established using SIP, the RTP (Real Time
Protocol) is used to handle voice data transfer in the packets form.
Refer RFC 1889 for more details.
NAT:
NAT (Network Address Translation-RFC 1631), is the translation of IP
address of a host in a packet. For example, the source address of an
outgoing packet, used within one network is changed to a different IP
address known within another network.
• One network is designated the inside network and the other is the
outside. Typically, a company maps its local inside network
addresses to one or more global outside IP addresses and unmaps
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the global IP addresses on incoming packets back into local IP
addresses. This helps ensure security since each outgoing or
incoming request must go through a translation process that also
offers the opportunity to qualify or authenticate the request or
match it to a previous request.
• NAT also conserves on the number of global IP addresses that a
company needs and it lets the company use a single IP address
in its communication with the world.
STUN: (RFC 3489)
Note:
STUN is used if there is a NAT router between SETU VP236 and SIP
Server (from ITSP).
• STUN stands for Simple Traversal of UDP over NAT. It is a
protocol, which enables an SETU VP236 to detect the presence
and type of NAT behind which the phone is placed. An SETU
VP236 that supports STUN can intelligently modify the private IP
address and port in its SIP/SDP message by using the NAT
mapped public IP address and port through a series of STUN
queries against a STUN server located on the public Internet.
• This will allow SIP signaling and RTP media to successfully
traverse a NAT without requiring any configuration changes on the
NAT. STUN presents a working solution for most NATs that are not
symmetric NAT, e.g., most of the SOHO routers have nonsymmetric NAT and in this case, it is OK to use STUN. However,
STUN does NOT work with symmetric NAT and if your routers have
built-in symmetric NAT, do not use STUN.
• A STUN server can help facilitate traversing through most NATs,
except for symmetric NATs.
Outbound Proxy
VoIP service provider can host a SIP outbound proxy server to handle
all of the SETU VP236’s VoIP traffic. User has to enable this feature if
VoIP service provider has a SIP outbound server to handle voice calls.
This allows the SETU VP236 to work with any type of NAT router and
eliminates the need of STUN.
SIP Call flow:
A simple example of how SIP requests and responses are sent will
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help understanding SIP and RTP.
When A calls B, the steps for the set up and tear down of a SIP call is
explained as below:
A
B
1. INVITE
2. Ringing
3. OK
4. ACK
5. Voice Data-RTP
6. BYE
7. OK
1. A sends SIP INVITE request to B. This message is an invitation to
B to participate in a SIP telephone call.
2. B sends a response that it is ringing.
3. A answers the call. B responds it by ‘OK’.
4. B also answers the call and A sends acknowledgment by sending
‘ACK’ message.
5. Now, voice session starts between A and B in form of RTP (Real
Time Protocol).
6. After the talk, A hangs up and sends a BYE request.
7. Finally, B sends the reply by sending ‘OK’ response and the call
is terminated.
Important Voice Parameters:
Jitter Buffer:
• The clarity of voice heard from the called party is also distorted due
to non-uniform delay introduced by the Telephony network known
as ‘Jitter’.
• In order to reduce the effect of ‘jitter’, some sort of ‘filters’ are used
in the voice-channel called ‘Jitter Buffer’. The SETU VP236 has
built-in adaptive buffer that helps to smooth out variations in delay
(jitter) for voice traffic. This will ensure good voice quality. It is not
programmable.
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Voice Compression:
• Voice compression is a technique to reduce the band width of voice
channel, using suitable codec or Coder-Decoder. The codec
converts voice signals from analog form to digital signals and vice
versa. It uses the compression alongwith, CELP.
• Compression-decompression is a voice compressiondecompression algorithm that defines the rate of speech
compression, quality of decompressed speech and processing
power requirements. It is called companding.
• Codec of suitable bandwidth is selected during the speech
reception or transmission, to minimize the digital voice channel
bandwidth.
QoS:
Quality of Service, QoS is the capability of a network to provide better
service to selected network traffic over various technologies, including
Ethernet and 802.1 networks, and IP-routed networks. SETU VP236
supports type of Service Tagging and Differentiated Services Tagging
as mentioned below: (This allows SETU VP236 to tag voice frames so
they can be prioritized over the network).
1. Type of Service (ToS)
2. DiffServe
ToS:
• Network traffic can be classified by setting the ToS values at the
data source (at the SETU VP236) so that a server can decide best
method of delivery, such as the least cost, fastest route, etc.
• This parameter allows you to configure Type of Service (ToS) bits
by specifying the precedence and delay of audio and signaling
IP packets.
DiffServe:
DiffServe is differentiated Services, which defines a Class of Service
model that makes packets so that they receive per hope treatment at
DiffServe-compliant network services along the route, based on the
application types and traffic flow. Packets are marked with DiffServe
Code Point (DSCP) indicating the level of service desired.
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Relevant Topic:
1. Glossary 179
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Web Jeeves
Introduction
• As, the set up of SOHO and Enterprise is normally equipped with
LAN and Router on the Internet line, it is very convenient if the
device has the capability to be programmed via HTML-based
interface. The Matrix SETU VP236 has this provision.
• Matrix SETU VP236 needs Web browser IE6 with SP2 (Service
Pack) patch or Mozilla Fire Fox installed on the PC. Description of
following features related to web will help in programming the
system.
IP Address
• An IP address (Internet Protocol address) is a unique number,
similar in concept to a telephone number, used by network devices
(computers, time-servers, some telephones).
• It is attached to a network to refer to each other when sending
information through a LAN or a WAN or the Internet for example.
• An example of IP address is 192.168.001.151. Converting a
number address to a more human-readable form called a domain
address is done via the Domain Name System.
• The Internet Protocol (IP) knows each logical host interface by a
number, the so-called IP address. On any given network, this
number must be unique among all the host interfaces that
communicate through this network.
• Users of the Internet are sometimes given a host name in addition
to their numerical IP address by their Internet service provider.
• For all programs that utilize the TCP/IP protocol, the sender IP
address and destination IP address are required in order to
establish communication and send data.
Sub Net
• It is a mechanism that is used to split a network into a number of
smaller sub networks. It can be used to reduce traffic on each sub
network by confining traffic to only the sub networks for which it is
intended, thereby eliminating issues of associated congestion on
other sub networks and reducing congestion in the network as a
whole.
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• Makes entire network more manageable.
• Each sub network functions as though it was an independent
network, keeping local traffic local, and forwarding traffic to another
sub network only if the address of the data is external to the sub
network. Such decisions are made on the basis of routing-tables
contained within the various routers, with each table comprising an
IP address table.
• Subnet is a portion of the network, which may be a physically
independent network, which shares a network address with other
portions of the network and is distinguished by a subnet number.
• The first octet of the IP address shows the network address, the
second one shows the subnet number and the last two shows the
host part.
Default Gateway
• A default gateway is a node on a computer network that serves as
an access point to another network.
• In enterprises, the gateway is the computer that routes the traffic
from a workstation to the outside network that is serving the Web
pages.
• In homes, the gateway is the ITSP that connects the user to the
internet.
• A default gateway is used by a host when an IP packet’s
destination address belongs to someplace outside the local subnet.
• The default gateway address is usually an interface belonging to
the LAN’s border router.
Domain Name System (DNS)
• DNS is a system that stores information about host names and
domain names in a type of distributed database on networks, such
as the Internet.
• Of the many types of information that can be stored, most
importantly it provides a physical location (IP address) for each
domain name, and lists the mail exchange servers accepting e-mail
for each domain.
• The DNS provides a vital service on the Internet as it allows the
transmission of technical information in a user friendly way.
• While computers and network hardware work with IP addresses
to perform tasks such as addressing and routing, humans generally
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find it easier to work with hostnames and domain names (such as
www.example.com) in URLs and e-mail addresses.
• The DNS therefore mediates between the needs and preferences
of humans and of software.
Dynamic Host Configuration Protocol (DHCP)
• The DHCP is a client-server networking protocol.
• A DHCP server provides configuration parameters specific to the
DHCP client host requesting, generally, information required by the
host to participate on the Internet network.
• DHCP also provides a mechanism for allocation of IP addresses to
hosts.
How to configure SETU VP236?
To start basic VoIP configuration first enter the default LAN IP address
of the SETU VP236, obtained from SE in the ‘Address’ bar of the
Web-browser screen as explained below:
• Launch the Web browser as explained below:
• Double Click on Internet Explorer icon.
Or
Click on Start Æ Programs Æ Internet Explorer
• Enter IP Address at Address Bar after confirming from SE. Let us
say IP Address 192.168.1.200 is entered at Address Bar on
opening page of Microsoft Internet Explorer.
• SETU VP236 displays Welcome page with a field for entering
‘Password’.
Parameter
Password
Value/Default
Up to '4' digits
(0-9), 1234
Description
Enter the default password
‘1234’ to access the system
• Click Login after entering valid password.
• If the user enters a wrong password then system will pop up a
window which will prompt him as, “Password is invalid, please
relogin” If user clicks on “log In” enters correct password and
press ‘Enter’, he will get ‘Welcome’ Page.
• To log out, click on ‘Logout”. You will find a thanks page asking
for Login Password.
• Enter the password if you want to work with SETU VP236.
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MATRIX SETU VP236
Welcome
User Address
Login expires if idle for
xxx.xxx.xxx.xxx
10 Minutes
xxx.xxx.xxx.xxx : Address of the computer from where the
user has logged in.
• Login Expiry Time: This is fixed (maximum) time for which user
can keep a page without programming any parameter in it. If this
time is exceeded, he cannot submit current page. He has to login
again.
• All the above fields are non-editable. The user can now go to any
of the screens by clicking on the links provided at the left of the
screen.
• Main Menu is displayed on the left side of Welcome Page (without
Password field).
• The ‘Main Menu List’ shows all the important parameters that are
required to be programmed, to make the SETU VP236 work as
expected.
• Click ……from the ‘Main Menu’.
Common screen command buttons: (General Note for all the
screens)
Note that at the end of the every screen, options are provided for
really applying these changes to the SETU VP236. This is conveyed
by ‘Submit’ and ‘Default All’:
Button
Meaning
Submit
Default All
Click, to save changes to VP236
Click, to set all parameter on the
screen to factory default
Links for following features are visible on the Welcome Page:
• Auto Configuration
• Call Logs
• Change Password
• Daylight Saving Time Adjustments
• Default IP Phone
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•
•
•
•
•
•
•
•
•
•
•
•
Keys Programming
Network Parameters
OG Call Route Selection
Peer-to-Peer Configuration
Peer-to-Peer Table
Phone Book
SIP Parameters
Software Upgrade
Status
System Parameters
Time Setting
User Setting
Screen Summary:
Link
Functions
Auto
This is configure the Phone with specific MAC
configuration
Address number by the ITSP
Call Logs
This shows record of various types of calls like,
IC, OG, Conference, Rejected calls etc
Change
This screen is used to change Keypad unlock
Password
and Config Phone Passwords for security
reasons
Daylight Saving This is to program new start and end times as
per the country’s DST rule.
Time
Adjustment
Default IP
This is to set again all parameters to factory set
Phone
values.
Keys
This page is used to assign important feature
Programming with a specific key with or without LED
Network
This is to configure Network port parameters
Parameters
OG Call Route This is to select the SIP1 or SIP2 Trunk for call
selection
routing
Peer to Peer
This is for programming the configuration
Configuration
parameters.
Peer to Peer
This is for programming Peer to Peer call Table
Call Table
with Number and Address
Phone Book
This is for programming phone-book entries
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SIP
Parameters
Software
Upgrade
Status
System
Parameters
Time Setting
User Setting
This screen is used for configuring SIP settings
like IP Address, Server Address etc for SIP1,
SIP2
This is to upgrade the software for any new or
current features.
This is a brief status of important parameters like
Network port parameters etc
This for programming parameters like, CPTG,
Ring, volume setting etc
Use this screen to configure system’s time to
synchronies with the time server
This is to set some features like, Call Forward,
Auto Answer etc.
How to make changes in feature?
In order to make changes, click on any corresponding feature, from
the list of the Web page and then refer related topic.
Example:
To use Phone Book feature for speed dialing.
• After entering suitable IP Address in address field the Welcome
Page will look as:
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• After log-in, the following page “Login Status Page’ displays the
time after which log-in will expire as shown below:
• Click on “Phone Book’ as shown:
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• Enter in the first Index, the name for which you want to dial a
number as shown below:
• Enter for the same index, the number of the person to be called:
• Similarly enter other Index and click on “Submit”.
• The phone book can be used now.
• Other features can be configured in the same way.
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Relevant Topics:
1. Change Password 73
2. Know Your SETU VP236 17
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Section 3: Appendices
Matrix
Appendix A: Glossary
CELP
It is the Code Excited Linear Prediction compression. It is the
compression algorithm used in low bit-rate voice encoding. Used in
ITU-T Recommendations G.728, G.729, G.723.
DHCP
It is the Dynamic Host Configuration Protocol. It provides a
mechanism for allocating IP addresses dynamically so that addresses
can be reused when hosts no longer need them.
DNS
It is the Domain Name System. System used on the Internet for
translating names of network nodes into addresses.
Firewall
Router or access server or several routers or access servers,
designated as a buffer between any connected public networks and a
private network. A firewall router uses access lists and other methods
to ensure the security of the private network.
TFTP
File Transfer Protocol - A protocol used to transfer files over a TCP/IP
network.
G.723
It is an ITU standard for speech codecs that uses the ADPCM method
and provides toll quality audio of 20 and 40 Kbps.
G.729
Describes CELP compression where voice is coded into 8-kbps
streams. There are two variations of this standard (G.729 and G.729
Annex A) that differ mainly in computational complexity; both provide
speech quality similar to 32-kbps ADPCM. It is described in the ITU-T
standard in its G-series recommendations.
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IP
It is the ‘Internet Protocol’, Network layer protocol in the TCP/IP stack
offering a connectionless internetwork-service. IP provides features
for addressing, type-of-service specification, fragmentation and
reassembly and security. It is defined in RFC 791.
MAC
Media Access Control Address-The unique address that a
manufacturer assigns to each networking device.
• MAC is Media Access Control address in the form of 48-bit
number, which is unique to the LAN NIC (network interface card).
• It is programmed into the card at the time of manufacture. IEEE
registration authority administers MAC address scheme for all
LANs which conform to IEEE, 802stds. Including both Ethernet
and token ring.
• Consists of two parts: 24-bit company id (manufacturer ID0 and
24-bit Extension id (board id)
• Destination and source MAC names are contained in the header
of the LAN packet and are used by various devices like hubs,
bridges.
• A VoIP Service Provider will typically have its subscribers register
with the Service Provider’s VoIP service server before starting a
subscriber’s service. Often, an ITSP will require the registration of
the MAC addresses of any devices directly connected to their
network.
SDP
It is the Session Definition Protocol, an IETF protocol for the definition
of Multimedia Services. SDP messages can be part of SGCP and
MGCP messages.
TCP
It is the Transmission Control Protocol, a connection-oriented
transport layer protocol that provides reliable full-duplex data
transmission. TCP is part of the TCP/IP protocol stack.
UAS
It is the User agent server (or user agent), a server application that
contacts the user when a SIP request is received and then returns a
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response on behalf of the user. The response accepts, rejects or
redirects the request.
UDP
User Datagram Protocol-A network protocol for transmitting data that
does not require acknowledgment from the recipient of the data that is
sent.
VoIP
It is the Voice over IP, the capability to carry normal telephony-style
voice over an IP-based Internet with POTS-like functionality, reliability,
and voice quality. VoIP enables a router to carry voice traffic (for
example, telephone calls and faxes) over an IP network. In VoIP, the
DSP segments the voice signal into frames, which then are coupled in
groups of two and stored in voice packets.
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Appendix B: Product Specifications
Configuration and Capacity:
• Headset Port : 1 (Audio Jack), 1 (MIC)
• Handset Port : 1 (RJ11)
• Network Port : 1 (RJ45) (PoE Supported)
• PC Port
: 1 (RJ45)
• DC Power
: 1 (Power Jack)
VoIP and Networking Protocol:
• SIP (RFC 3261, Ver.2)
• IPv4-Internet Protocol Version 4 (RFC791)
• ARP-Address Resolution Protocol (RFC 0826)
• ICMP-Internet Control Message Protocol (RFC0792)
• TCP-Transmission Control Protocol (RFC793)
• UDP-User Datagram Protocol (RFC768)
• RTP-Real Time Protocol (RFC1889)
• RTCP-Real Time Control Protocol (RFC1889)
• SNTP-Simple Network Time Protocol
• PPP-Point-to-Point Protocol (RFC 1661)
• DNS Protocol (RFC 2136)
• Outbound Proxy
• STUN
• NATKeep Alive
• VLAN Tag
Voice Functionalities:
• Maximum two SIP Accounts can be used simultaneously
• Maximum four Call Appearances are supported
• DTMF Options: Inband, RTP (RFC2833), SIP (INFO)
• Call Progress Tone Generation: Select as per the country
• QoS: RTP QoS and SIP QoS using DiffServ and ToS
(Precedence)
• Full Duplex Audio
• Echo Cancellation
• Speech Volume Setting-Transmit and Receive
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Voice Codecs: PCM (a-Law and u-law), G.729, G.723-5.3, G.723-6.3,
G.726-16, G.726-24, G.722, G.726-32, G-726-40.
Telephony features:
• Voice Calls using SIP proxy and Voice calls without using SIP
proxy (Peer-to-Peer Calling)
• Call Waiting and Cancel Call Waiting
• Caller ID Blocking
• Call Forwarding
• 3-Party Conference
• Call Transfer
• Caller ID: Display
• Speed Dialing with Phone Book Feature
• Voice Mail Account: Two
• LCD Displays: Size : 2 line/6 line, with 24 characters and tilt facility.
Brightness, Contrast : Programmble
Time Settings:
• Synchronizing with specific Time Server
Provisioning, Administration and Maintenance:
• Auto Configuration
• Configuration Using Web Page, Phone-’Menu’ and Programmble
keys.
• LED Indications:
• Ringer LED
• Speaker LED
• Programmble keys with LED
• Software Upgrade using FTP
Security:
• Password Protected Administrator.
Power Supply:
External Adaptor
Power Consumption
: [email protected]. or through Power on
Ethernet.
: 10W (Typical)
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Environmental:
Operating Temperature
Operating Humidity
Storage Temperature
Storage Humidity
: -10°C to 50°C (140° to 122°F)
: 5-95% RH, Non-Condensing
: -40°C to 85°C (-40°F to 185°F)
: 0-95% RH, Non-Condensing
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Index
A
ACB Notification-refer Auto Call Back 42
Alphanumeric Key-refer Making a Call 94
Anonymous-refer Anonymous Call Rejection 39
Attended Transfer-refer Call Transfer 69
Authentication-refer SIP Parameters 128
Auto Answer Timer-refer Auto Answer 40
B
Backlight-See LCD Settings 92
Blind Transfer-refer Call Transfer 69
Brightness-refer LCD Settings 92
Busy Tone-refer Call Progress Tone Generation 65
C
Call Center Application-refer Auto Answer 40
Call Forward Timer-refer Call Forward see also Timers 52, 150
Call Hold Alert Timer-refer Call Hold 56
Call Route Selection-refer OG Call Route Selection 110
Call Waiting Tone (CCWT)-refer Call Progress Tone Generation 65
Conference-refer 3-Party Conference 37
Configuring Password-refer Change Password 73
Contrast-refer LCD Settings 92
Current Time-refer Time Settings 148
D
Default Password-refer Default SETU VP236 79
Delete All Contacts-refer Phone Book 117
Delete a Character-refer Know Your SETU VP236 17
Destination Address-See Peer-to-Peer Calling 113
DHCP-refer Network Port 102
Dial Pad-refer Know Your SETU VP236 17
Dial Tone-refer Call Progress Tone Generation 65
DND List-refer Do Not Disturb 83
DNS-refer Network Port 102
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DTMF Option-refer Peer-to-Peer Calling see also SIP Parameters
113, 128
E
End of Dialing-refer Making a Call see also System Parameters 94,
144
Error Tone-refer Call Progress Tone Generation 65
F
G
GMT-refer Time Settings 148
H
Hotline Timer-refer Hotline 85
I
Invalid Input-refer Call Appearances 49
IP Address-refer Network Port see also Receiving a Call
102, 119
K
Keypad Lock-Auto-refer Key Assignment 87
Keypad Lock-Manual-refer Key Assignment 87
Keypad Lock Timer-refer Key Assignment 87
L
LCD Display-refer Know Your SETU VP236 17
LCR-refer OG Call Route Selection 110
Local User’s Name-refer Peer-to-Peer Calling 113
M
MAC-refer Status 138
N
NAT-refer Network Port 102
NAT Type-refer Status 138
Navigation Key-refer Phone Book 117
Network Status-refer Menu 100
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Notification Display Timer-refer Notifications 108
Number of Call Appearances-refer System Parameters 144
NTP Server-refer Time Settings 148
O
OG Call Route-refer Dial Plan 81
OG/IC Call When Connected-refer Menu 100
Ordinal-See Daylight Saving Time 76
Outbound Proxy-refer SIP Parameters 128
Override Character-refer Know Your SETU VP236 17
P
P2P Calling-See Peer-to-Peer Calling 113
Phone Book Index Number-refer Key Assignment see also Speed Dial
87, 136
PPPoE-refer Network Port 102
Priority1: Remote Held Call-refer Rejecting a Call 122
Public IP-refer Status 138
Q
QoS-refer Network Port 102
R
Redial Key-refer Last Number Redial 91
Registrar Server-refer SIP Parameters 128
Registration Last Fail-refer Status 138
Registration retry Count-refer Status 138
Remote Held Resuming Tone-refer Call Hold 56
Reset-refer Default SETU VP236 79
Retrieving Held Call-refer Call Hold 56
Ring Back Tone-refer Call Progress Tone Generation 65
Ringer Key-refer Receiving a Call 119
Ringer LED-refer Know Your SETU VP236 17
Ringer Tone Type-refer Rings 124
RTP Listening Port-See Network Port 102
S
SIP Account-refer SIP Parameters 128
SIP ID-refer SIP Parameters 128
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SIP1/SIP2 Key-refer Selective SIP Line 127
SIP Listening Port-See Network Port 102
Statistics-refer Status 138
STUN-refer Network Port see also SIP Parameters 102, 128
Subnet-refer Status 138
T
TFTP server-refer Auto Configuration 44
Time Zone-refer Time Settings 148
Transfer Notification Timer-refer Call Transfer 69
Transferee-refer Call Transfer 69
Transferor-refer Call Transfer 69
Transfer Target-refer Call Transfer 69
U
Up/Down Key-refer Search Contact 126
User Password-refer Change Password 73
V
VM Retrieval-refer Voice Mail 157
VM Subscription-refer Voice Mail 157
Voice Mail Account1-refer User Settings 152
Vocoder Preference-refer Peer-to-Peer Calling 113
see also SIP Parameters 128
Volume Settings-refer System Parameters 144
W
Waiting Call-refer Call Appearances 49
Web Page-refer Web Jeeves 168
Welcome Page-refer Web Jeeves 168
X
Y
Z
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